JP2935213B2 - Audio information transmission method - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a high-efficiency speech encoding / decoding apparatus (speech CODE) for compressing and transmitting speech information.
More specifically, the present invention relates to an audio information transmission system for transmitting a PB (push button) signal in the same manner as an audio signal.

[0002]

2. Description of the Related Art Conventionally, this kind of high efficiency voice CODEC
Are described in, for example, JP-A-63-86933,
As shown in Japanese Patent Publication No. 246535, it is used for a digital line, and in the digital line, in addition to a voice signal, a PB signal can be passed, and the transmission quality of the PB signal is guaranteed.

FIG. 5 shows a high-efficiency audio CODEC using a conventional audio information transmission system (Japanese Patent Laid-Open No. 63-86933,
JP-A-2-246535). In this system, the transmitting section T has a high-efficiency speech coder 1, a PB signal receiving circuit 2, and a multiplexer 4, and the receiving section R includes a separator 5, a high-efficiency speech decoder 6, and a PB signal generating circuit. 8 and a switch 9, and the transmission of the audio signal and the PB signal in the data format shown in FIG.

That is, an audio signal or a PB
When a signal is input, the high-efficiency speech coder 1 converts the input signal into a waveform coding algorithm (for example, ADPM, APC,
High-efficiency speech information compression is performed by a multi-pulse coding method or the like and an analysis / synthesis algorithm (for example, LPC, PARCOR, LSP method, etc.), and the result is input to the multiplexer 4 as speech coded data.

On the other hand, the PB signal receiving circuit 2 has a filter for detecting a PB signal from an input signal therein. The PB signal detected by the filter is converted into a PB character signal and input to the multiplexer 4. At the same time, the PB input detection signal is input to the multiplexer 4. The multiplexer 4 normally selects the speech coded data and outputs it to the digital line 10. When the PB input detection signal is input, the multiplexer 4 outputs the selected output to the PB.
It is designed to switch to a character signal.

Here, the transmission data output from the multiplexer 4 is, as shown in FIG.
ms), and there is a 1-bit determination bit (indicated by S) at the beginning of the frame. This determination bit indicates that the frame is audio signal data when S = 0, and S =
A value of 1 indicates that the frame is PB signal data.

When such data is input to the receiving unit R via the digital line 10, the separator 5 looks at the decision bit S in the received data, and when S = 0, the speech encoding in the frame is performed. The data is input to the high-efficiency speech decoder 6, and when S = 1, the PB character data in the frame is input to the PB signal generation circuit 8.

The high-efficiency speech decoder 6 receives the encoded speech data, decodes the encoded speech data into an original speech signal, and outputs it to the switch 9. When the PB character data is input, the PB signal generation circuit 8 generates a PB signal by decoding the PB character data into the original PB signal and outputs the PB signal to the switch 9, and at the same time, outputs the PB input detection signal to the switch 9. Output. The switch 9 is usually
Although the audio signal is selectively output, when the PB input detection signal is input, the selected output is switched to the PB signal.

The PB signal of the conventional high-efficiency audio CODEC is a Japanese NTT standard (a certain PB) shown in FIG.
Minimum 1 from signal input to next PB signal input
A time difference of 20 msec or more (PB tone: minimum 50 msec or more, pause: minimum 30 msec or more) is required. ) Based.

On the other hand, in the encoding method of the input signal (audio signal / PB signal), the input signal is sampled at the rising edge of the frame synchronization signal of 20 msec. For this reason, the transmission of the audio signal and the PB signal is performed in a frame unit of 20 msec.

However, the conventional high efficiency voice CODE
C transmits the voice signal and the PB signal on a transmission side in a frame unit of 20 msec. Therefore, if the PB signal is coded using the conventional method, the PB signal output is not necessarily the NTT standard of the PB signal shown in FIG. May be difficult to satisfy.

Here, as a prior art, for example, Japanese Unexamined Patent Publication No.
Japanese Patent Application Laid-Open No. 9-49093 (hereinafter referred to as prior art 1) discloses that a PB signal is generated using a microprocessor by using a microprocessor.
By performing OM reading, when reading out various types of signal data having different word lengths cyclically in a sampling cycle,
A "digital PB signal generation method" is disclosed, which has a simple circuit configuration and can promptly target a new PB signal even when a new PB signal is required due to addition of program data.

[0013]

As described above, in the conventional audio information transmission system, the audio signal and the PB signal are time-division multiplexed and transmitted in units of 20 msec on the transmission side by the audio CODEC. Therefore, there is a disadvantage that the encoded output of the PB signal may not always satisfy the NTT standard of the PB signal.

SUMMARY OF THE INVENTION The present invention has been made to solve the above-mentioned problem. In a digital line using a high-efficiency audio CODEC, a PB signal which satisfies the NTT standard of the PB signal and which can guarantee the quality in addition to the audio signal is used. Can be passed through,
An object of the present invention is to provide a voice information transmission method capable of stably realizing a PB service using a high-efficiency line.

Here, the prior art 1 discloses an invention of a digital PB signal generation system for reading a ROM using a microprocessor to generate a PB signal. There is no specific description suggesting the technical idea of the present invention, which converts the data into a format and converts it to PB audio at the transmission destination, thereby enabling PB signal transmission without adverse effects of encoding. Is based on a completely different technical idea.

[0016]

In order to achieve the above-mentioned object, a voice information transmission system according to the present invention provides a transmission unit which encodes a voice signal with high efficiency in frame units of a fixed period and outputs voice coded data. A non-speech encoding means for detecting a non-speech signal and encoding the non-speech signal in frame units; and a tone section of the non-speech signal encoded by the non-speech encoding means. And an in-band processing unit that expands the pause section by at least a predetermined number of frame periods and outputs expanded data, and outputs a detection signal by determining whether audio or non-speech is performed on a frame-by-frame basis. In the section where a voice signal is detected, the decompressed data and the other voice-encoded data are selected and multiplexed, and the multiplexed data is derived to a digital line as transmission data. A receiving unit that receives the transmission data transmitted via the digital line as reception data, a separator that separates the reception data into voice encoded data and decompressed data, Voice decoding means for decoding the voice-encoded data separated by the separator into a voice-decoded signal; and inputting the decompressed data separated by the separator, each frame is a tone section or a pause section. Frame mode detection means for determining whether the frame mode detection means determines the tone section mode, and inputs the decompressed data separated by the separator and decodes it into a non-speech decoded signal to determine the pause section mode A non-speech decoding means for stopping the output when the non-speech decoding means receives the non-speech decoding signal from the non-speech decoding means. The non-speech decoding signals sent when there is no input of the non-speech decoding signal is characterized in that so as to have a switch for transmitting an audio decoded signal of the speech decoding means.

[0017]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS An embodiment of the present invention will be described below in detail with reference to FIGS.

FIG. 1 is a block diagram showing the configuration of a high-efficiency audio CODEC using the audio information transmission system according to the present invention. In FIG. 1, the same parts as those in FIG. 5 are denoted by the same reference numerals.

In the transmitting section T, the high-efficiency speech encoder 1
Compresses and encodes the audio signal. PB signal receiving circuit 2
Has a function of detecting whether the input signal is a PB signal and a function of encoding the PB signal. The in-band processing circuit 3 has a function of converting the encoded PB signal into in-band.

When a speech signal or a PB signal is input from the transmitting side, the high-efficiency speech coder 1 converts the input signal into a waveform coding algorithm (for example, APC, multi-pulse coding, etc.) or an analysis and synthesis algorithm (for example, , LPC, P
ARCOR, LSP system, etc.) to perform high-efficiency audio information compression, and use the result as coded audio data.
To enter.

The PB receiving circuit 2 detects the input signal and
The signal is converted into PB character data, and the PB character data is input to the in-band processing circuit 3. The in-banding processing circuit 3 performs in-banding of PB character data (processing for converting the data into a data format in which a data discrimination bit and a PB tone detection code are provided in each data frame).

More specifically, the input signal is a PB signal (a PB tone of 50 msec and a pause of 70 msec) indicated by A in FIG. 3A, and this PB signal is shown in FIG.
When the sampling is performed at the rising edge of the frame synchronization signal shown in FIG. 3, the reproduced PB tone is 40 msec as indicated by A 'in FIG. FIG. 3 shows the input signal.
(A) PB signal indicated by B in the middle (PB tone of 90 msec and 3
When the PB signal is sampled at the rising edge of the frame synchronization signal shown in FIG. 3B, the reproduced PB pause is 20 msec as indicated by B 'in FIG. 3C.

As described above, when the input of the PB signal shown in FIG. 3A is sampled at the rising edge of the frame synchronization signal shown in FIG. 3B, the NT of the PB signal shown in FIG.
The T standard cannot be satisfied.

Therefore, as shown in FIGS. 4A to 4C, the PB tone is forcibly expanded to 60 msec or more, converted into PB character data shown in FIG. I do. Also, when switching from the PB tone to the PB pose, the PB pose is forcibly extended to 40 msec or more as shown in FIGS. The signal is converted and input to the multiplexer 4.

The multiplexer 4 normally sends the encoded voice data to the digital line 10, but when a PB signal detection signal is input, switches to PB character data and outputs the data to the digital line 10.

On the other hand, on the receiving unit R side, the separator 5 determines the received data, and if the received data is PB character data, inputs the PB character data to the frame mode detection circuit 7. If the received data is speech encoded data, the speech encoded data is input to the high-efficiency speech decoder 6.

The high-efficiency audio decoder 6 receives the encoded audio data, decodes the encoded audio data into an audio signal, and inputs the audio signal to the switch 9. When the PB character data is input, the frame mode detection circuit 7 determines whether or not the frame is a PB tone frame.
To enter. Then, when switching from the PB tone frame to the PB pause frame, the PB signal generation circuit 8 is reset, and the reproduction of the PB tone is stopped.

When PB character data is input, PB signal generation circuit 8 converts the PB character data to PB character data.
The signal is decoded into a signal and input to the switch 9. This switch 9
Normally outputs an audio signal, but switches to a PB signal when a PB signal is input.

Therefore, the high-efficiency voice C having the above configuration
When the transmitting unit T detects the PB signal, the ODEC sets the tone period and the pause period of the encoded PB signal to at least a predetermined number of frames (PB signal N
(Corresponding to the TT standard)
B, the section in which the PB signal is detected is set to the PB expanded output, and the other is configured to output the speech coded data. The receiving unit R detects the frame mode of the PB signal and converts the PB signal to the output. Decode and output audio signal and PB
By switching and transmitting the signal, it is possible to pass a PB signal which satisfies the PB signal NTT standard and whose quality can be guaranteed, in addition to the audio signal. This makes it possible to stably provide a PB service using a highly efficient line.

Although the transmission of the PB signal has been described in the above embodiment, the present invention can be similarly applied to other non-voice signals. In addition, various modifications can be made without departing from the spirit of the present invention.

[0031]

As described above, according to the present invention, in a digital line using a high-efficiency voice CODEC,
In addition to audio signals, PB signals that meet the NTT standard and whose quality can be guaranteed can be passed, and an audio information transmission method that can stably realize a PB service using a highly efficient line can be provided.

[Brief description of the drawings]

FIG. 1 is a block diagram showing a configuration of a high-efficiency audio CODEC using an audio information transmission system according to the present invention.

FIG. 2 is a timing chart showing a transmission data format of the embodiment.

FIG. 3 is a timing chart showing a timing of inputting a PB signal and a state of reproduced data when an in-band processing is not performed in the embodiment.

FIG. 4 is a timing chart showing a timing of input of a PB signal and a state of reproduced data when in-band processing is performed in the embodiment.

FIG. 5 shows a highly efficient voice C using a conventional voice information transmission system.
FIG. 3 is a block diagram illustrating a configuration of an ODEC.

FIG. 6 is a timing chart showing a transmission data format according to a conventional audio information transmission method.

FIG. 7 is a timing chart showing the NTT standard of the PB signal.

[Explanation of symbols]

 REFERENCE SIGNS LIST 1 high-efficiency speech encoder 2 PB signal receiving circuit 3 in-band processing circuit 4 multiplexer 5 demultiplexer 6 high-efficiency speech decoder 7 frame mode detection circuit 8 PB signal generation circuit 9 switch 10 digital line

Claims (3)

(57) [Claims] A transmitting unit configured to efficiently encode an audio signal in a frame unit having a predetermined period and output encoded audio data; Non-speech encoding means for encoding the speech signal, and expanding the tone section and the pause section of the non-speech signal encoded by the non-speech encoding means by at least a predetermined number of frames, respectively, and outputting decompressed data And in-band processing means for determining whether the signal is speech or non-speech on a frame-by-frame basis and outputting a detection signal. And a multiplexer that multiplexes the selected data and derives the multiplexed data to a digital line as transmission data, and the transmission data transmitted through the digital line. A receiving unit for receiving as received data, a separator for separating the received data into audio encoded data and decompressed data, and an audio decoder for decoding the audio encoded data separated by the separator into an audio decoded signal Converting means, frame mode detecting means for inputting the decompressed data separated by the separator to determine whether each frame is a tone section or a pause section, and determining that the frame mode detecting means is a tone section mode. When the decompressed data is input to the decompressed data, the decompressed data is decoded into a non-speech decoded signal, and the output is stopped when the pause section mode is determined. When a non-speech decoded signal is input, the non-speech decoded signal obtained by the non-speech decoding means is sent to an output line. A switch for transmitting a decoded audio signal of the audio decoding means. 2. A data format in which a data discrimination bit is provided in a data frame divided for each frame period as transmission data from the transmission unit to the reception unit, and a data discrimination in each data frame by in-band processing. 2. The voice information transmission system according to claim 1, wherein a data format provided with bits and a PB tone detection code is used. 3. The audio information transmission system according to claim 1, wherein said non-audio signal is a push button signal.