Data processing: speech signal processing – linguistics – language – Audio signal bandwidth compression or expansion
Reexamination Certificate
2001-05-08
2004-08-24
To, Doris H. (Department: 2655)
Data processing: speech signal processing, linguistics, language
Audio signal bandwidth compression or expansion
C704S201000, C704S504000, C704S503000
Reexamination Certificate
active
06782367
ABSTRACT:
TECHNOLOGICAL FIELD
The invention concerns generally the field of encoding and decoding a signal to be transmitted over a telecommunication connection. Especially the invention concerns the procedures of changing the signal bandwidth of such a signal during the course of the telecommunication connection.
BACKGROUND OF THE INVENTION
FIG. 1
illustrates the general principle of transmitting speech from a first terminal to a second terminal in a digital cellular radio network. In the first terminal
100
there is a series connection of a microphone
101
, a speech encoder
102
, a channel encoder
103
, a modulator
104
and a radio transmitter
105
. In a first base station
110
there is a series connection of a radio receiver
111
, a demodulator
112
, a channel decoder
113
and a line transmitter
114
. From the first base station
110
to a second base station
120
there is a network connection
115
. The second base station
110
comprises a series connection of a line receiver
121
, a channel encoder
122
, a modulator
123
and a radio transmitter
124
. In a second terminal
130
there is a series connection of a radio receiver
131
, a demodulator
132
, a channel decoder
133
, a speech decoder
134
and a loudspeaker
135
.
The speech encoder
102
in the transmitting terminal
100
converts the analogue speech signal that comes from the microphone
101
into a digital signal by applying a certain speech encoding scheme. The channel encoder
103
adds redundancy to the digital signal in order to enhance its robustness against corrupting effects at the radio interface. The channel decoder
113
removes at least partly the channel decoding, because wired connections through the network
115
are much more reliable than radio connections and excessive channel coding would only consume transmission capacity in the network. A corresponding pair of channel encoding
122
and channel decoding
133
exists around the second radio interface. The speech decoder
134
reconverts the digital speech signal into analog by applying a procedure that is an inverse of the above-mentioned speech encoding scheme. The principles described above are easily generalized to the transmission of arbitrary information between terminals by replacing the microphone
101
with a generic data source, the speech encoder
102
with a source encoder, the speech decoder
134
with a corresponding decoder and the loudspeaker
135
with a generic data sink.
An encoding and decoding unit is usually referred to as a codec. The specifications of conventional digital cellular radio systems like the original GSM (Global System for Mobile telecommunications) typically define speech (or source) codecs that have a constant output bit-rate and that handle a speech (or source) signal the bandwidth of which is constant. Depending on the bandwidth the conventional speech codecs have been designated as either narrowband or wideband codecs. For example the so-called RPE-LTP full-rate speech codec described in the GSM standard number GSM 06.10 is a narrowband speech codec the bandwidth of which is approximately 3.5 kHz. Its bit-rate in speech coding is 13 kbit/s and in channel coding 9.8 kbit/s which together makes 22.8 kbit/s. Exemplary wideband speech codecs are those standardized by the ITU (International Telecommunication Union) under the designations G.722-64, G.722-56 and G.722-48. Their speech coding bit-rates are 64, 56 and 48 kbit/s respectively, and their bandwidth is approximately 7 kHz.
Recent proposals for enhancements to the known arrangements in speech (or source) coding include the concept of AMR or Adaptive MultiRate coding. The idea is to keep the bit (or symbol) rate at the output of the channel encoder
103
constant but to allow the roles of the speech encoder
102
and the channel encoder
103
to change in generating the constant bit-rate. The input bandwidth of the speech encoder is constant (in GSM AMR, the same 3.5 kHz as in the basic GSM speech codec mentioned above), but if the speech encoder is allowed to use more bits per time unit, better audible quality can be achieved. Using a larger portion of the available bit-rate for speech coding is only possible on condition that the corruptive effects of noise and interference of the moment are not too bad. At the receiving end the AMR concept means that the bit (or symbol) rate at the input of the channel decoder
133
is constant, but the amount of redundancy removed in the channel decoder and correspondingly the amount of digital information per time unit available for reconstructing the original analog speech signal in the speech decoder
134
may vary.
At the priority date of the present patent application the known AMR speech coding principle is going to be adopted in standardizing a wideband or 7 kHz speech codec for future use within the GSM frameworks. It is possible that in the near future there will be communication devices in use which have two selectable speech (or source) bandwidths: 3.5 kHz and 7 kHz. It is also possible that even more speech (or source) bandwidths will be defined. The bandwidths can be associated with the use of completely different codecs or they may represent just certain modes of operation, known as the codec modes or just modes, of the speech encoding and decoding arrangements. The application of the AMR principle means that a future speech (or source) codec may have both a selectable bandwidth and a changing bit-rate, where the latter is associated with different levels of error protection through different distributions of the available gross bit-rate between speech (or source) coding and channel coding.
FIG. 2
illustrates in more detail the contents of the speech encoder block
102
in a transmitting mobile station and the speech decoder block
134
in a receiving mobile station in a known exemplary case where two different speech bandwidths have been defined. Here the concepts of encoding and decoding are understood in a wide sense so that e.g. A/D and D/A conversions are parts thereof. The A/D converter
201
in the encoder
102
is coupled to a switching block
202
both directly and through a downsampling block
203
. The output of the switching block
203
is coupled to a speech encoder proper
204
which is capable of handling both a wideband and a narrowband input signal. The communication channel
210
between the output of the speech encoder proper
204
and the input of a corresponding speech decoder proper
220
in the speech decoder block
134
comprises generally e.g. all channel encoding/decoding and transmitting/receiving arrangements. The speech decoder proper
220
is capable of decoding both wideband and narrowband speech signals, and the output thereof is coupled to a switching block
221
both directly and through an upsampling block
222
. The output of the switching block
221
is coupled to a speech synthesizer and D/A converter
223
.
The A/D converter
201
in the encoder block
102
and the D/A converter
223
in the decoder block
134
both handle a sampling rate that is high enough for the widest defined speech bandwidth. The downsampling block
203
reduces the sampling rate of the sample stream produced by the A/D converter
201
to a lower level by puncturing, filtering or interpolating, and the upsampling block
222
inflates the sampling rate of the sample stream produced by the speech decoder proper
220
to a higher level by some calculational means. As a response to a bandwidth change command the speech encoder
204
and decoder
220
switch to encoding and decoding procedures that correspond to the new bandwidth, and simultaneously the switching blocks
203
and
221
select either the direct couplings (in the case of wider bandwidth) or those going through the downsampling block
203
and upsampling block
222
(in the case of narrower bandwidth). Multiple bandwidths can be achieved by programming the speech encoder
204
and decoder
220
for multiple bandwidths and by providing multiple parallel downsampling blocks in the transmitting station and upsampling block
Mikkola Hannu
Rotola-Pukkila Jani
Vainio Janne
Nokia Mobile Phones Ltd.
Perman & Green LLP
To Doris H.
Vo Huyen X
LandOfFree
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