KR0144935B1 - Coding and decoding apparatus for bit rate - Google Patents

Abstract

In the encoding and decoding apparatus capable of adjusting the bit rate according to the present invention, the input audio signal and the video signal are separately encoded at various bit rates according to a user's selection, and the bit stream formed by the encoder and the encoder for forming the encoded bit stream are audio. And a decoder for classifying and restoring the signal and the video signal.

Therefore, the stored or transmitted data can be selected and encoded or decoded at one of several predetermined bit rates, and the bit rate can be adjusted separately for the audio signal or the video signal according to the user's selection. Mutual bit rate adjustment is possible.

Description

Encoding and decoding device with adjustable bit rate

1 is a block diagram of an encoding apparatus capable of adjusting a bit rate according to the present invention.

2 is a block diagram of a decoding apparatus capable of adjusting a bit rate according to the present invention.

3 is a detailed block diagram of the encoding apparatus shown in FIG.

4 is a detailed block diagram of the decoding apparatus shown in FIG.

5A to 5E are diagrams showing examples of bit use.

6A and 6B show examples of quantization and quantization steps.

7A to 7C show processing unit frames for each mode.

The present invention relates to an encoding and decoding apparatus capable of adjusting a bit rate, and more particularly, to an encoding and decoding apparatus capable of selectively adjusting a bit rate by a user.

Audio and video equipment is a device that allows a user to obtain voice information or image information when needed, and is a device that obtains information through the ears and eyes among the five senses, which are channels of human information acquisition.

The signals present in nature generally exist in analog form, a continuous signal form. These analog signals have a problem of easily deterioration when used in a reproducing apparatus or a transmission apparatus, a problem of difficult compensation for a possible error signal, and a problem of increasing the size of a system when a system is configured. These problems can be solved by transforming analog signals into digital signals through sampling and quantization.

Since the information obtained changes over time, data must be obtained at every time interval. In consideration of this, a sampling interval suitable for real-time processing of the reference data is determined when arranging an arbitrary system. When the system is configured, these points must be satisfied in real time. An example of this fact is that a film of a movie in a movie theater must be repeated at least several times a second to be recognized as continuous information by human eyes or sampling of a CD. It is related to setting the frequency to 44.1 kHz.

Data in the time domain are obtained by the determined sampling interval, and the sampled values may be quantized and represented as a digital signal. When the signal is processed in the digital method, the sound quality is improved compared to the analog method, while the data amount is increased compared to the analog method.

Data compression is performed to reduce the impact of this increase in data.

If data compression is not performed, transmission of video information or audio information is impossible because the channel cannot provide the bandwidth required in many cases. For example, in a gray-scale image having a resolution of 1000 x 1000 lines in the case of an image, when each pixel is composed of 8 bits and represented with different intensity of 256 gray levels for each pixel, transmission of the image is performed without compression. The number of bits used will reach 8 million bits. In a typical telephone line, 9600 bits can be transmitted per second, so it takes more than 10 minutes to transmit this image. As another example, in the case of audio sampled at 44.1 kHz, each sampled information consists of 16 bits. When the 65536 gray level is used, the number of bits for one channel is about 700 k and about 73 seconds when using a general telephone line. There is an impossible problem.

In such data transmission, the bit rate used for transmission is determined according to the characteristics of the transmission channel, and the bit rate for real time processing is determined. The determination of the transport channel led to the development of devices for optimally reconstructing each transmitted data and encoding devices to use the data rate as effectively as possible. In order to effectively use transmission and storage limits, many studies on data compression have been conducted.

For this reason, as data compression, various methods such as waveform coding, transform coding, band division coding, fractal coding, predictive coding, and motion compensation encoding have been developed, and efforts to create a standardization scheme for recording and transmitting video and audio are under ISO. Was started in 1989 by ISC / IEC JTC1 / SC29 / WG11 (MPEG).

However, in the related art, since the maximum bit rate that can be transmitted is fixed when a transmission channel is determined, it is possible to design a specific reproducing apparatus that is optimally effective at the transmission rate. Not developed. That is, when certain information is transmitted, it is impossible for the user to arbitrarily select and restore the bit rate of the data to be transmitted, and only a specific reproduction apparatus can restore it.

In addition, when video and audio information are stored on an optical disc, it is impossible to selectively adjust the bit rate for the video and audio information. That is, it was impossible to control image quality and sound quality while adjusting the bit rate allocated to the image information and the bit rate allocated to the audio information among the total bit rates.

Accordingly, an object of the present invention is to provide an apparatus for encoding or decoding by selecting one of several predetermined bit rates of stored data or transmitted data in order to solve the above problems.

In order to achieve the above object, an encoding and decoding apparatus capable of adjusting a bit rate according to the present invention includes an encoder for separately encoding an input audio signal and a video signal at various bit rates according to a user's selection, and forming an encoded bitstream; And a decoder for classifying and restoring the bitstream formed by the encoder into an audio signal and a video signal.

Hereinafter, with reference to the accompanying drawings will be described in detail with respect to the present invention.

1 is a block diagram of a coding apparatus capable of adjusting a bit rate according to the present invention, wherein a video signal processor 11, an audio signal processor 12, and first and second encoders 13 and 14 are shown. It is composed of a control unit 15 and a bit packing unit 16 and a detailed block diagram is shown in FIG.

2 is a block diagram of a decoding apparatus capable of adjusting the bit rate according to the present invention. The bit unpacking unit 21, the audio signal processor 22, the video signal processor 23, the first and second decoders 24 and 25 are shown in FIG. ) And a control unit 26, the detailed block diagram of which is shown in FIG.

5a to 5e show examples of bit use, FIG. 5a shows standard bits, FIG. 5b shows audio signals highlighted, FIG. 5c shows video signals highlighted, and FIG. 5d shows video signals only. 5e shows a case where only an audio signal is present.

6A and 6B are diagrams showing examples of quantization and quantization steps.

7A to 7C show processing unit frames for each mode, and FIG. 7A shows a case where a fixed unitary bit rate is increased, and FIG. 7B shows a bit rate adjustment using a fixed maximum unit. Each case of bit rate increase that is not shown is shown. Here, SNC is synchronization, nf is normalization factor, bai is bit allocation information, ba2 is bit allocation 2, and 2n is 0≤2n≤2k.

Next, the operation of the present invention will be described with reference to FIGS. 1 to 7.

According to the present invention, an audio signal and a video signal are encoded by an encoder to form an encoded bitstream, and the decoder classifies and restores the bitstream into audio and video signals. As shown in FIGS. 1 and 3, the encoding apparatus includes a time / frequency converter, a controller for controlling a bit rate, a bit rate representation method, a time domain processing or a frequency domain processing, a first encoder for the smallest bit rate, Decoding unit for the case where the bit rate is small, error calculation unit for calculating the difference between the decoded signal and the original signal, the second bit stream from the error signal by selecting one of modes 1, 2, 3 according to the bit rate representation method selected from the control The secondary bitstream forming unit for forming the first bitstream and the second encoding unit for forming the final bitstream using the secondary bitstream and the bitstream having the smallest bit rate.

Meanwhile, the decoding apparatus decodes primary bits by dividing the input signal into primary bits and secondary bits by synchronizing the input signal as shown in FIGS. 2 and 4, and uses the additional information of the primary bits to control the secondary bits. A bit forming unit is configured to read and decode according to a bit rate according to a control signal, and implements scalability by combining information of secondary bits with information of secondary bits.

First, referring to the encoding apparatus, audio, video, or a / v is selected to determine the type of data to be stored. When the type of data to be stored is determined, the bit rate that can be stored is determined in consideration of the storage capacity of the channel, and the frame size is determined according to the bit rate. For example, if the bit rate (Vbit) used for video is set to k, the process of calculating the bit rate (Abit) used for audio using the bit rate (AVbit) used for audio / video determined by the characteristics of the channel is Vbit. If it is less than or equal to AVbit, it can be represented as Abit = AVbit-Vbit.

That is, in the case of FIG. 5a, if you want to listen to the audio with higher quality by increasing the bit rate of the audio signal, decrease the Vbit value and use it as in FIG. 5b.In contrast, if you want to see the higher quality of the video signal by increasing the bit rate of the video signal, Vbit. Increase the to use as shown in Figure 5c. In the extreme case, the entire AVbit is used for video or audio, which is the same as either 5d or 5e.

Data compression is performed according to the predetermined minimum bit rates Abit (min) and Vbit (min) at the size of the frame determined according to the type of data previously determined. The predetermined minimum bit rate is used to create the base restoration signal to provide scalability. The remaining number of bits other than the predetermined minimum bit rate is used to provide various bit rates for providing scalability.

In other words, for audio,

Abit2 = Abit-Abit (min)

For video,

Vbit2 = Vbit-Vbit (min)

The Abit2, Vbit2, computed by is used to provide the bitrate to provide scalability.

There may be many ways to use, store and restore the remaining bits at various bit rates. This method depends on whether the bit rate increases to a fixed value, to a non-fixed value, to increase the bit rate by one unit as the query increases, or to consider the bit rate when decoding extra information. There can be three same ways.

Method 1: increase bit rate by fixed unit

Method 2: Bit rate adjustment using fixed maximum unit

Method 3: Unfixed Bit Rate Increase

Looking at each method in more detail as follows.

Method 1: Increase bit rate by fixed unit.

Depending on whether the value used for data processing is the value of the frequency domain or the time domain, the processing is as follows. In the case of the value of the frequency domain, for example, if there are k frequency domain values according to the resolution of the frequency domain, the process of reading 2k bits more by increasing the sound quality one by one and reading 2k bits less by decreasing the sound quality one by one Allows adjustment of sound quality. In the case of time-domain values, the sound quality is increased by reading 2k bits more for k time-domain values, and lowered by reading less. At this time, if the size of the frame for the audio signal or the video signal is determined and the size of the frame is M1 and the size of the minimum bit rate for each signal is M2, the extra bits available for bit rate adjustment are M1-M2. If the value of the resolution of each signal is k, the more usable values for each signal are M3 = (M1-M2) / k. Dividing k by 2 calculates the number of units for the secondary bitstream.

The reason why the sound quality is adjusted according to the increase or decrease of the number of beats is that the processing of reducing the quantization error can be performed as the number of bits increases, and the closer to the original sound as the quantization error decreases. The reason why 2k bits are used in changing the bit rate is to consider the processing of +1, * Δ, 0, -1 * Δ. Here, Δ is a value corresponding to half of the delta when the quantization step is called delta.

Here, as in the case of the DPCM, the original signal is restored by using the difference from the previous signal. 6 shows an example in which a quantization step of a signal and a quantizer is operated. As the value of M increases, the number of bits increases to M + Δ, M-Δ, and M, which is closer to the original signal. For example, if the quantized values have values of 0, 10, 20, 30, 40, ..., 250, if the original signal is 77, the quantized value is 80. In this case, Δ is 5, but if you increase the number of bits by 2 and process for -Δ, the newly quantized value is 75, which has an error of -2 less than the error of +3, which is 80. Come close. In this case, the new Δ becomes 2.5, and if the number of bits is increased by two more, the quantization value is 77.5, which can reduce the error to 0.5 and get closer to the original sound.

Method 2: Bit rate adjustment using fixed maximum unit

Using a fixed bit-rate increment of one unit is a hassle to consider for each or every minus sign. It solves this problem and reduces the redundancy of the code expression by expressing the case where the existing bits represent three cases (+ Δ, 0, -Δ) with two bits for the case of 2EXP (k) -1 with the bit k. One way is to use a fixed maximum number of units to efficiently use data. If the size of the frame for the audio signal or the video signal is determined as in the method 1, and then the size of the frame is M1 and the size of the lowest bit rate for each signal is M2, the extra bits available for bit rate adjustment are M1-M2. Becomes If the value of the resolution of each signal is k, the more usable values for each signal are M3 = (M1-M2) / k. That is, there may be M3 more bits for each signal, thereby reducing the quantization error with Δ * 2 EXP (M3-1) as much as possible. In this case, bit rate adjustment is as follows. As an example of the method 1, when the value of 77 becomes 80, if M3 bits are 4, for example, subtracting 1 bit for sign representation and expressing the difference with the remaining 3 bits can express an error of -3 as in '011'. have.

In this case, when the user wants to improve the sound quality, the value represented by the extra bit M3 is read to influence the output value by comparison.

That is, when L = -3

Case 1: 0 | L | ≤ 1,

Case 2: 1 | L | ≤ 2,

Case 3: 2 | L | ≤ 3,

By reducing the error value in turn can improve the sound quality or image quality.

For example, in case 2, when a user enjoys voice data or image data, and then changes to case 1 as required, the user can lower the sound quality or the image quality to perform the listening process. In this case, after reading the bits for the extended part, the modified values are calculated according to the user's request.

Method 3: Unfixed Bit Rate Increase

In the case of the fixed bit rate increase and decrease in the method 1 and the method 2, in order to efficiently allocate bits according to the increase or decrease of the bits, the audio is processed based on the threshold band. By processing based on the critical band, the effect of error due to quantization processing is localized, and the processing more suitable for human hearing psychology is possible. In this case, if the frequency resolution is k in the scheme 1 or scheme 2, unlike the bit allocation by 2k or M3 * k, the bits are allocated to the unit blocks by dividing the frequency domain or the time domain into unit blocks used for processing. . For example, in the case of audio data having a threshold band of 24 bands, bits can be efficiently used by having a flag unit for assigning or not allocating bits for each band.

For example, if the quality can be improved by using one more bit for only the processing block 3, the 24 bits of the flag portion have a value equal to '0 0 1 0 0 ... 0', and only the processing of the third processing block is more detailed. In addition, the flag unit may have a change in sound quality or image quality by giving values for + Δ, 0, and -Δ for the third processing block signal.

The processing unit frame of the audio portion or the video portion is formed by adding the information compressed by the minimum bit rate determined above to the information by the remaining number of bits other than the minimum bit rate.

As a result of the processing, the shape of the processing unit frame for each mode is shown in Figs. 7A to 7C. This processing unit frame is used to store data on disk or tape or to transmit data wirelessly or by wire.

Meanwhile, the process of reading or receiving stored data or transmitted data is performed by the decoding apparatus of FIGS. 2 and 4. In order to recover, first, signals to be read to the processing buffer are different depending on whether they are audio or video signals, and the amount of signals in the buffer is adjusted according to the bit rate. Depending on the difference between the minimum bit rate and the desired bit rate, a value is calculated to provide various bit rates with values similar to the calculated values.

When the signals are filled in the processing standby buffer, decoding processing according to a predetermined bit rate is performed, and values for providing various bit rates provided by the selected mode are decoded and modified according to the result of the decoding process according to the minimum bit rate. It has a change in sound quality or image quality according to the variation of the bit rate. The processing example by this method is as follows.

When the signal calculated by the minimum bit rate is m * delta, if the signal reconstructed by the second bit is n, the signal at this bit rate becomes m * delta + (delta / 2) * n. Also, delta is 10 as one unit, and m is 80 and n is -1. In the example, 80 + (10/2) * (-1) is 75, and if the bit is increased by one more unit, m is Since 75, n is +1 and delta is 5, it can be seen that 75+ (5/2) * (+ 1) becomes 77.5.

As described above, in the video and audio encoding and decoding apparatus capable of adjusting the bit rate according to the present invention, it is possible to separately adjust the bit rate of the audio signal or the video signal according to the user's selection, and the bit rate control between the audio signal and the video signal is It is possible.

In addition, by simultaneously recording and reproducing or transmitting the various bit rates, the user can not only select the bit rate if necessary, but also express the data efficiently by preventing the redundancy of the data representation by expressing the various bit rates in various modes. In addition, there is an advantage that can be effectively transmitted according to the characteristics of the transmission line during transmission using the bitstream for various bit rates.

Claims (6)

An encoding apparatus for forming a bitstream by encoding an input audio signal and a video signal at a predetermined bit rate, the encoding device comprising: a predetermined minimum bit rate determined for each of the minimum bits necessary for representing the input audio signal and the video signal. A first encoder configured to encode the audio signal and the video signal, respectively, to form a first bitstream; a decoder to decode the first bitstream signal encoded by the first encoder; an audio signal decoded by the decoder; An error calculator which calculates an error signal between the video signal and the input audio signal and the video signal, respectively; differently set bit rates for the audio signal and the video signal according to a predetermined selected bit rate, and the audio signal and the video signal Encode an error signal for A second encoder for forming a second bit stream; And a formatter for forming a final bitstream by combining the first bitstream generated by the first encoder and the second bitstream generated by the second encoder. A first encoder configured to form a first bitstream by encoding the audio signal and the video signal according to a predetermined minimum bit rate, respectively, for the minimum bit required for the representation of the input audio signal and the video signal; A second encoder which encodes an audio signal and a video signal obtained by decoding a bitstream signal and an error signal between the input audio signal and the video signal according to a bit rate selected according to a predetermined setting mode to form a second bit stream; And a formatter configured to add a first bitstream generated by the first encoder and a second bitstream generated by the second encoder to form a final bitstream. A decoding apparatus for restoring a bitstream transmitted from an encoding apparatus encoding a bit rate to form a bitstream, into an original audio signal and a video signal, comprising: an input for separating and storing an audio signal and a video signal from the transmitted bitstream A first decoder to decode a first bitstream included in the audio signal and a video signal; a second decoder to decode a second bitstream included in the audio signal and the video signal, respectively; And an output buffer for transforming the value decoded by the first decoder according to the value decoded by the second decoder. The encoding apparatus according to claim 1, wherein the bit rate of the second encoder is determined according to a bit rate of one of the audio signal and the video signal. 4. The encoding apparatus of claim 3, wherein the bit rate of the audio signal is determined by subtracting the minimum video bit rate from the bit rates of the entire audio signal and the video signal. 4. The encoding apparatus of claim 3, wherein the bit rate of the video signal is determined by subtracting the minimum audio bit rate from the bit rates of the entire audio signal and the video signal. An encoding and decoding apparatus comprising an encoder for encoding and transmitting an input audio signal and a video signal at a predetermined bit rate, and a decoder for reconstructing a signal transmitted from the encoder into an original audio signal and a video signal. A first encoder configured to form a first bitstream by encoding the audio signal and the video signal according to a predetermined minimum bit rate, respectively, for a minimum bit necessary for representing the input audio signal and the video signal; A decoder configured to decode the first bitstream signal encoded by the encoder; an error calculator configured to calculate the error signals of the audio and video signals and the audio input and video signals decoded by the decoder; The audio signal according to a bit rate selected according to a setting mode A second encoder for forming a second bit stream by coding each of the error signal for the video signal; And a formatter configured to combine the first bitstream generated by the first encoder and the second bitstream generated by the second encoder to form a final bitstream. The decoder includes audio from the transmitted bitstream. An input buffer configured to separately store a signal and a video signal; a first decoder configured to decode a first bitstream included in the audio signal and the video signal, respectively; decoding a second bitstream included in the audio signal and the video signal, respectively; A second decoding unit; And an output buffer for transforming and outputting the value decoded by the first decoder according to the value decoded by the second decoder.