Note : Les descriptions sont présentées dans la langue officielle dans laquelle elles ont été soumises.
CA 02293165 2000-04-06
S. A. Benno 1
METHOD FOR TRANSMITTING DATA IN
WIRELESS SPEECH CHANNELS
Background of the Invention
1. Field of the Invention
The present invention relates to telecommunications; more particularly, to
transmitting
data in wireless speech channels.
2. Description of the Prior Art
A voice encoder/de':oder (vocoder) is used to compress voice signals so as to
reduce
the transmission bandwidth over a communications channel. By reducing the
bandwidth per
call, it becomes possible to place more calls over the same channel. There
exists a class of
vocoders known as code excited lineax prediction (CELP) vocoders. In these
vocoders, the
speech is modeled by a series of filters. The parameters to these filters can
be transmitted with
much fewer bits than the original speech. It is also necessary to transmit the
input (or
excitation) to these filters in order to reconstruct the original speech.
Because it would require
too much bandwidth to transmit the excitation directly, a crude approximation
is made by
replacing the excitation by a. few non-zero pulses. The locations of these
pulses can be
transmitted using very few trits and this crude approximation to the original
excitation is
adequate to reproduce high quality speech. The excitation is represented by a
fixed codebook
contribution and an associated gain. Also the quasi-periodicity found in
speech is represented
by an adaptive codebook output and an associated gain. The fixed codebook
output and its
associated gain, the adaptive: codebook output and its associated gain, and
filter parameters
(also known as linear predictive coder parameters) are transmitted to
represent the encoded
speech signal.
The vocoders were initially designed to compress speech by modeling its
characteristics
and transmitting the parameters of that model in much fewer bits than
transmitting the speech
itself. As wireless phones become more commonplace, people are increasingly
expecting to use
them for the same range of non-speech applications as they have used
traditional landline
phones, such as accessing voice mail and receiving call waiting tones.
Recently, the FCC has
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S. A. Benno 1
mandated that text-telephones for the hearing impaired (TTY/TDD) work with
digital cellular
phones. The problem with non-speech applications is that they do not fit the
vocoder's speech
model. When non-speech signals are passed through the vocoder, the decoded
result it not
always acceptable. The problem is further exacerbated by the fact that
wireless phones operate
in an error prone environment. In order to recover from transmission errors,
the vocoder
depends on a speech model to recover from random errors. Once again, non-
speech signals do
not match this model and so the reconstruction is inadequate.
Summary of the Invention
The present invention sends information in the bits allocated to one or both
of the
codebooks' output by setting; the gain for the corresponding codebook to zero.
By setting the
gain to zero, the codebook output will not be interpreted by the receiving
vocoder. In this way,
it is possible to transmit additional information in a way that is totally
transparent to the
vocoder. Applications for this technique of sending "secret" messages include,
but is not
limited to, transmitting parameters for generating non-speech signals. As an
example,
information to generate call waiting tones, DTMF, or TTY/T'DD characters can
be clandestinely
embedded in the compressed bit stream so that these non-speech tones can be
regenerated.
Brief Description of the Drawings
FIG. 1 is a block diagram of a typical vocoder;
FIG. 2 illustrates the major functions of encoder 14 of vocoder 10; and
FIG. 3 is a functional block diagram of decoder 20 of vocoder 10.
Detailed Description
FIG. 1 illustrates a block diagram of a typical vocoder. Vocoder 10 receives
digitized
speech on input 12. The digitized speech is an analog speech signal that has
been passed
through an analog to digitized converter, and has been broken into frames
where each frame is
typically on the order of 20 nnilliseconds. The signal at input 12 is passed
to encoder section 14
which encodes the speech so as decrease the amount of bandwidth used to
transmit the speech.
The encoded speech is made available at output 16. 'The encoded speech is
received by the
decode section of a similar vocoder at the other end of a communication
channel. The decoder
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S. A. Benno 1
3
at the other end of the communication channel is similar or identical to the
decoder portion of
vocoder 10. Encoded speech is received by vocoder 10 through input 18, and is
passed to
decoder section 20. Decoder section 20 uses the encoded signals received from
the transmitting
vocoder to produce digitized speech at output 22.
Vocoders are well known in the communications arts. For example, vocoders are
described in "Speech and audio coding for wireless and network applications,"
edited by Bishnu
S. Atal, Vladimir Cuperman, and Allen Gersho, 1993, by Kluwer Academic
Publishers.
Vocoders are widely available and manufactured by companies such as Qualcomm
Incorporated of San Diego, California, and Lucent Technologies Inc., of Murray
Hill, New
Jersey.
FIG. 2 illustrates the major functions of encoder 14 of vocoder 10. A
digitized speech
signal is received at input 12, and is passed to linear predictive coder 40.
Linear predictive
coder 40 performs a linear predictive analysis of the incoming speech once per
frame. Linear
predictive analysis is well l~:nown in the art and produces a linear
predictive synthesis model of
the vocal tract based on the input speech signal. The linear predictive
parameters or
coefficients describing this model are transmitted as part of the encoded
speech signal through
output 16. Coder 40 uses this model to produce a residual speech signal which
represents the
excitation that the model uses to reproduce the input speech signal. The
residual speech signal
is made available at output 42. The residual speech from output 42 is provided
to input 48 of
open-loop pitch search unit 50 to an input of adaptive codebook unit 72 and to
fixed codebook
unit 82.
Impulse response unit 60 receives the linear predictive parameters from coder
40 and
generates the impulse response of the model generated in coder 40. This
impulse response is
used in the adaptive and fixed codebook units.
Open loop pitch search unit 50 uses the residual speech signal from coder 40
to model
its pitch and provides a pitch, or what is commonly called the pitch period or
pitch delay signal,
at output 52. The pitch delay signal from output 52 and the impulse response
signal from
output 64 of impulse response unit 60 are received by input 70 of adaptive
codebook unit 72.
Adaptive codebook unit 72 produces a pitch gain output and a pitch index
output which become
part of encoded speech output 16 of vocoder 10. Output 74 of adaptive codebook
72 also
provides the pitch gain and :pitch index signals to input 80 of fixed codebook
unit 82.
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S. A. Benno 1
4
Additionally, adaptive codebook 72 provides an excitation signal and an
adaptive codebook
target signal to input 80.
The adaptive codebook 72 produces its outputs using the digitized speech
signal from
input 12 and the residual speech signal produced by linear predictive coder
40. Adaptive
:i codebook 72 uses the digitized speech signal and linear predictive coder
40's residual speech
signal to form an adaptive c;odebook target signal. The adaptive codebook
target signal is used
as an input to fixed codebook 82, and as an input to a computation that
produces the pitch gain,
pitch index and excitation outputs of adaptive codebook unit 72. Additionally,
the adaptive
codebook target signal, the pitch delay signal from open loop pitch search
unit 50, and the
1 (1 impulse response from impulse response unit 60 are used to produced the
pitch index, the pitch
gain and excitation signals which are passed to fixed codebook unit 82. The
manner in which
these signals are computed is well known in the vocoder art.
Fixed codebook 82 uses the inputs received from input 80 to produce a fixed
gain
output and a fixed index output which are used as part of the encoded speech
at output 16. The
15 fixed codebook unit attempts to model the stochastic part of the linear
predictive coder 40's
residual speech signal. A target for the fixed codebook search is produced by
determining the
error between the current adaptive codebook target signal and the residual
speech signal. The
fixed codebook search produces the fixed gain and fixed index signal for
excitation pulses so as
to minimize this error. The manner in which the fixed gain and fixed index
signals are
20 computed using the outputs from adaptive codebook unit 72 are well known in
the vocoder art.
Switches 90 and 92 are used to send data in place of the bits that are used to
send the
fixed codebook output and the adaptive codebook output, respectively. When the
contacts of the
switches are in position "A'", the associated codebook output is replaced by
data or other
information and the associal:ed codebook gain is set to zero or substantially
zero. As a result,
25 the scaled codebook output ~or excitation produced at a receiver will be
zero or substantially
zero and therefore will not have an adverse affect on the filter being used by
the receiving
vocoder to model the speech that is normally transmitted.
FIG. 3 illustrates a functional block diagram of decoder 20 of vocoder 10.
Encoded
speech signals are received at input 18 of encoder 20. The encoded speech
signals are received
30 by decoder 100. Decoder 100 produces fixed and adaptive code vectors
corresponding to the
fixed index and pitch index aignals, respectively. These code vectors are
passed to the
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S. A. Benno 1
excitation construction portion of unit 110 along with the pitch gain and the
fixed gain signals.
The pitch gain signal is used to scale the adaptive vector which was produced
using the pitch
index signal, and the fixed ;gain signal is used to scale the fixed vector
which was obtained
using the fixed index signali. Decoder 100 passes the linear predictive code
parameters to the
_'c filter or model synthesis section of unit 110. Unit 110 then uses the
scaled vectors to excite the
f lter that is synthesized using the linear predictive coefficients produced
by linear predictive
coder 40, and produces an output signal which is representative of the
digitized speech
originally received at input 12. Optionally, post filter 120 may be used to
shape the spectrum of
the digitized speech signal that is produced at output 20.
1 Ci When data rather than speech information is being transmitted, the pitch
index
(adaptive codebook output) and/or the fixed index (the fixed codebook output)
are used to
receive the data. The affect of non-data signals on the filter synthesize by
unit 110 are
eliminated because the gain value associated with the pitch or code index is
zero.
The functional block diagrams can be implemented in various forms. Each block
can be
15 implemented individually using microprocessors or microcomputers, or they
can be
implemented using a single microprocessor or microcomputer. It is also
possible to implement
each or all of the functional blocks using programmable digital signal
processing devices or
specialized devices received from the aforementioned manufacturers or other
semiconductor
manufacturers.
