WO2008049311A1 - Procédé, système et appareil pour transmettre le flux de code encodé du bruit de fond - Google Patents

Procédé, système et appareil pour transmettre le flux de code encodé du bruit de fond Download PDF

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Publication number
WO2008049311A1
WO2008049311A1 PCT/CN2007/002680 CN2007002680W WO2008049311A1 WO 2008049311 A1 WO2008049311 A1 WO 2008049311A1 CN 2007002680 W CN2007002680 W CN 2007002680W WO 2008049311 A1 WO2008049311 A1 WO 2008049311A1
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Prior art keywords
code stream
layer
rate
background noise
coding
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PCT/CN2007/002680
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English (en)
French (fr)
Inventor
Libin Zhang
Jun Zhang
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Huawei Technologies Co., Ltd.
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Publication of WO2008049311A1 publication Critical patent/WO2008049311A1/zh

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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/012—Comfort noise or silence coding

Definitions

  • the present invention relates to the field of voice communications, and in particular, to a method, system and apparatus for transmitting coded stream of background noise. Background technique
  • the processing of speech is mainly done by a speech codec.
  • the original speech codec is fixed rate, that is, each speech encoder has only a fixed rate.
  • the higher rate encoding algorithm can ensure the encoding quality more easily, but the communication channel resources are larger.
  • the lower rate encoding algorithm occupies less communication channel resources, but does not. It is too easy to guarantee the quality of the code.
  • VAD Voice Activity Detector
  • the speech coding rate in the same speech coder is not limited to only the coding rate for speech and the coding rate for background noise.
  • Such an encoder can provide multiple encoding rates for speech coding, known as a variable rate speech coder. Since the variable rate speech coder can dynamically adjust the coding rate, so that the voice communication system can flexibly trade between the synthesized speech quality and the system capacity, the variable rate speech coder has been rapidly developed.
  • voice encoders not only process voice signals, but also process packets. A variety of music signals, including music, which require different encoding rates, so variable rate speech encoders are even more important.
  • the encoder selects the encoding rate.
  • One is based on the characteristics of the speech signal itself, that is, the source control.
  • the selection of the rate by using the VAD technology is a simple example of source control; the other is based on the status of the communication channel. , that is, channel control.
  • a typical example of source control is that the speech encoder encodes the speech signal at different rates depending on whether the speech signal is voiced or unvoiced, whether the voiced sound is stable or the like.
  • a typical example of channel control is that the coder requires the encoder to encode the voice signal at different rates according to the condition of the channel. If the channel condition is not good, it is busy. To save bandwidth, the voice signal is required to be used at a lower rate. Encoding, otherwise if the channel condition is good, not busy, and the bandwidth is sufficient, it is required to encode the speech signal at a higher rate to obtain higher synthesized speech quality.
  • the latest speech encoder adopts a new encoding rate forming method.
  • the basis of the encoding rate forming method is that the speech encoded code stream output by the encoder is encapsulated into a frame in a layered manner, that is, the encoded code stream is composed of
  • the core layer is composed of one or more enhancement layers, each layer is a set of coding parameters, and each set of coding parameters corresponds to a coded bit number.
  • the coded code stream of a certain voice is composed of a core layer and 11 enhancement layers, as shown in FIG. Show:
  • Layer 1 is a core layer occupying 8 kbits of coded bits
  • layer 2 is a narrowband enhancement layer occupying 4 kbits of coded bits
  • layers 3 to 12 are wideband enhancement layers occupying 2 kbits of coded bits, if the frame rate is transmitted per second.
  • the corresponding coding rates are 8kbit/s, 12kbit/s, 14kbit/s, 16kbit/s, 18kbit/s, 20kbit/s, 22kbit/s, 24kbit/s, 26kbit/s, 28kbit/s, 30kbit/s and 32kbit/s. Based on this rate formation mode, each rate is backward compatible.
  • layer 1 is reserved, that is, decoded at a rate of 8 kbit/s.
  • the channel does not have to indicate which rate the encoder uses to encode, and the encoder encodes the speech signal at the highest rate of 32 kbit Zs, and then encodes the codes.
  • the stream is transmitted to the communication channel layer by layer, and the channel determines the number of layers to be received according to the current channel condition.
  • the channel condition is good, the channel capacity is large, and it is not busy, then the channel will receive all the coded streams, that is, receive all the coding layers; if the channel condition is poor, busy, and the available capacity is small, then the channel The receiving capability is limited, only the first part of these coding layers is received, and the latter part is discarded, such as dropping layers 7 ⁇ 12, thus leaving the part below layer 6, ie receiving layer 1-6, as shown in picture 2.
  • the speech coder encodes each speech signal at a rate of 32 kbit/s without considering the specific conditions of the channel, and hierarchically transmits the 32 kbit/s encoded code stream to the communication channel.
  • the channel selects the actual transmission rate according to the actual channel condition. Therefore, the channel control mode selected by this rate simplifies the interaction process, and makes the channel control the coding rate more flexible.
  • the principle of synthesis of background noise is the same as that of speech synthesis.
  • the principle of speech synthesis is:
  • This model is also used in the synthesis of background noise, so the content of the characteristic parameters describing the background noise and the mute characteristic transmitted in the background noise coded stream is basically the same as the characteristic parameters in the speech coded code stream, which is the synthesis filter during signal synthesis. Parameters and excitation parameters.
  • the synthesis filter parameters are mainly Line Spectium Frequence (LSF) quantization parameters
  • the excitation signal parameters include: pitch delay parameters, pitch gain parameters, fixed codebook parameters, and fixed codebook gain parameters.
  • LSF Line Spectium Frequence
  • the number of quantization bits and the quantization form of these parameters are different; the same encoder, if it contains multiple rates, at different rates, due to the different emphasis of the characteristics of the description signal, the number of quantization bits of the coding parameters And the form of quantification is also different.
  • the background noise coding parameters describe the background noise characteristics. Since the excitation signal of the background noise can be regarded as a simple random sequence of noise, these sequences can be simply generated by the random noise generation module at the codec end. Then use energy parameters to control the amplitude of these sequences Degree, the final excitation signal can be generated, so the excitation signal characteristic parameter can be simply represented by the energy parameter without further description by other characteristic parameters, so in the background noise coded code stream, the excitation parameter is The energy parameter of the current background noise frame, which is different from the speech frame; the same as the speech frame, the synthesis filter parameter in the background noise coded stream is also the line spectrum frequency LSF quantization parameter, but the specific quantization method is different.
  • the background noise coded code stream is essentially a low-rate "voice" coded code stream.
  • the rate of speech or audio signals is higher in each variable rate speech coder, and when the background noise is processed, since the background noise coded frame describes only the background noise of the current speech communication environment, it is not the main body of communication, so the encoder When designing your own background noise coded frame, it is relatively simple, and the rate is low. Only some simple information of background noise is encoded. The background noise recovered at the decoding end is only a simulation of the background noise of the coded end. Very precise. Therefore, it can also be understood that the encoding method for background noise encoding is actually a simple low-rate speech encoding method.
  • variable rate speech coder there is a coding rate specifically for background noise coding, but each encoder encodes only one rate for background noise.
  • each encoder When encoding background noise, each encoder only uses The same fixed noise coding rate, that is, the same type of noise coded frame, encodes the background noise.
  • the bandwidth resources are sufficient, the most important thing for the two parties is the quality of the communication.
  • the high-quality coding mode is required to encode the voice, and the high-quality coding method is needed for the background when the call is not in use.
  • the noise is encoded.
  • the former point is easy to satisfy because variable rate speech coder has a large number of speech coding rates available for the encoder to choose from, but the latter point cannot be satisfied, because no matter how abundant the bandwidth is, when encoding the background noise, the code stream only corresponds to one type.
  • the coding rate so due to the limitation of the coding rate, although the bandwidth allows, the communication channel can only directly receive the background noise coded stream sent by the sender, and send the coded stream to the receiver, and the receiver can only use the code.
  • the rate is decoded, and the communication channel and the receiving end cannot flexibly select and control the coding quality. Summary of the invention Embodiments of the present invention provide a method, system, and apparatus for encoding code stream transmission of background noise, which are used to solve the problem that the receiving end cannot flexibly select and control the encoding quality in the prior art.
  • the transmitting end sends the coded code stream of the background noise to the receiving end at a coding rate.
  • the coded layer included in the coded stream includes a core layer and one or more enhancement layers, each layer includes a set of coding parameters, each set of coding parameters corresponding to a coded bit number, and the coding rate is included according to the coded code stream.
  • the number of coded bits corresponding to the layer is determined, and the core layer includes coding parameters that enable the background to recover the background noise independently when the receiving end decodes the coding rate corresponding to the core layer, where the enhancement layer includes the previous coding layer of the layer. Encoding parameters of the encoding effect;
  • the receiving end After receiving the encoded code stream of background noise, the receiving end determines a decoding layer, determines a decoding rate according to the decoding layer, and decodes the encoded code stream at the rate.
  • An encoder an encoded code stream for transmitting background noise to the decoder at a coding rate
  • the coded code stream comprising an encoding layer including a core layer and one or more enhancement layers, each layer including a set of coding parameters, each group of codes
  • the parameter corresponds to a coded bit number
  • the coding rate is determined according to the coded code stream including the number of coded bits corresponding to each layer
  • the core layer includes the background that enables the receiving end to decode at the coding rate corresponding to the core layer.
  • a coding parameter that is recovered by the noise alone, and the enhancement layer includes coding parameters that enhance the coding effect of the previous coding layer of the layer;
  • a decoder an encoded code stream for receiving background noise, determining a decoding layer according to a current processing capability or an application environment of the receiving end, determining a decoding rate according to the decoding layer, and decoding the encoded code stream at the rate.
  • An encoder is connected to a decoder, where the encoder includes: a first processing unit, configured to send an encoded code stream of background noise to a decoder at an encoding rate, where the encoded code stream includes
  • the coding layer includes a core layer and more than one enhancement layer, each layer containing a set of coding parameters, each set of coding parameters corresponding to a coded bit number, the coding rate is determined according to the coded code stream including the number of coded bits corresponding to each layer, and the core layer includes the receiving end corresponding to the core layer
  • the enhancement layer includes an encoding parameter that enhances the encoding effect of the previous encoding layer of the layer.
  • a decoder is provided in the embodiment of the present invention, and is connected to an encoder.
  • the decoder includes: a second processing unit, configured to receive an encoded code stream of background noise, and determine a decoding layer according to a current processing capability or an application environment of the receiving end. Determining a decoding rate based on the decoding layer and decoding the encoded code stream at the rate.
  • the transmitting end sends the coded code stream of the background noise to the communication channel at a coding rate, where the coded code stream includes a core layer and one or more enhancement layers, and the coded code stream arrives at the communication.
  • the communication channel can select the number of layers of the received coded stream according to the channel condition, and send the received coded stream to the receiving end, and the receiving end can also determine the decoding rate according to the current processing capability or the application environment and at the rate.
  • the encoded code stream is decoded, so the communication channel and the receiving end can flexibly select and control the coding quality.
  • FIG. 1 is a schematic structural diagram of a speech coded code stream in the prior art
  • FIG. 2 is a schematic diagram of controlling a layer of a received speech coded code stream by a communication channel in the prior art
  • FIG. 3 is a schematic flowchart of a method in an embodiment of the present invention.
  • FIG. 4 is a schematic structural diagram of a background noise coded code stream according to an embodiment of the present invention.
  • FIG. 5 is a schematic diagram of receiving a background noise coded code stream when a communication channel is in good condition according to an embodiment of the present invention
  • FIG. 6 is a schematic diagram of receiving a background noise coded code stream when a communication channel is in poor condition according to an embodiment of the present invention
  • FIG. 8 is a bit allocation diagram of a background noise coded frame according to an embodiment of the present invention
  • FIG. 9 is a schematic structural diagram of a system according to an embodiment of the present invention.
  • FIG. 10 is a schematic structural diagram of an encoder according to an embodiment of the present invention.
  • FIG. 11 is a schematic structural diagram of a decoder in an embodiment of the present invention. detailed description
  • the embodiment of the present invention provides a processing method for encoding code stream transmission of background noise.
  • the core idea of the method is: The transmitting end sends a background noise coded code stream including a core layer and one or more enhancement layers to a communication channel, and the communication channel is The channel condition selects the number of layers of the received coded stream, and sends the received coded code stream to the receiving end, and the receiving end determines the decoding rate according to the number of layers included in the coded code stream and decodes the coded code stream at the rate. .
  • a specific implementation process of a method for encoding code stream transmission of background noise includes the following steps:
  • Step 301
  • the encoded code stream is encapsulated into a frame in a layered manner, and the encapsulated encoded code stream includes a coding layer including a core layer and one or more enhancement layers, and each layer includes a set of coding parameters, where
  • the coding parameters included in the core layer enable the receiving end to recover the background noise alone when decoding at the coding rate corresponding to the core layer, and the coding parameters included in the enhancement layer can enhance the coding effect of the previous coding layer of the layer.
  • each group of coding parameters corresponds to a coded bit number
  • the coding rate corresponding to the core layer is the number of coded bits corresponding to the core layer multiplied by the frame rate, where the frame rate is the number of frames transmitted per second.
  • the transmitting end determines a coding rate according to the number of coding bits corresponding to each layer included in the coded stream, and the determining method is the sum of the number of coded bits corresponding to each layer included in the coded stream, multiplied by the frame rate, and then the transmitting end
  • the encoded code stream of background noise is transmitted to the communication channel at the encoding rate.
  • the communication channel determines the number of layers receiving the coded stream according to the current channel condition, such as the busyness level, that is, determines which layer of the background noise coded stream is received, and The encoded code stream is received in accordance with the number of layers. If the channel conditions are poor, it is very busy. If the bandwidth is tight, the enhancement layer will be discarded and only the core layer will be received. If the channel condition is good, not busy, and the bandwidth is sufficient, the channel can receive the enhancement layer in addition to the core layer. The communication channel then transmits the received encoded code stream to the receiving end.
  • the current channel condition such as the busyness level
  • Step 303
  • the receiving end receives the encoded code stream sent by the communication channel, first determines the number of decoding layers, and after determining the number of decoding layers, the decoding layer is determined accordingly. For example, if the number of decoding layers determined by the receiving end is 3, then the decoding layer is The core layer, the enhancement layer 1 and the enhancement layer 2, if the number of decoding layers determined by the receiving end is 5, the decoding layer is the core layer, the enhancement layer 1, the enhancement layer 2, the enhancement layer 3, and the enhancement layer 4. Then, the receiving end determines the decoding rate based on the determined decoding layer and decodes the received encoded code stream at the rate.
  • the background noise coded stream received by the receiving end only includes the core layer
  • the number of decoding layers is 1, and the decoding layer is the core layer
  • the decoding rate is the coding rate corresponding to the core layer, that is, the number of coded bits corresponding to the core layer. Multiplying the rate obtained by the frame rate; if the background noise coded stream received by the receiving end includes the core layer and the enhancement layer, the receiving end determines the number of decoding layers according to its own condition, and the decoding layer is determined accordingly, and the decoding rate is the decoding layer.
  • the corresponding coding rate is the sum of the number of coded bits corresponding to the decoding layer multiplied by the frame rate.
  • the current processing power determines the number of decoding layers. If the current processing capability of the receiving end is small, the number of decoding layers selected by the receiving end is small, and the corresponding decoding rate is small. If the current processing capability of the receiving end is strong, the number of decoding layers selected by the receiving end is Larger, the corresponding decoding rate is larger. In this way, the receiving end can flexibly select and control the decoding rate according to its current processing capability.
  • the application environment of the receiving end In different application environments, the required coding quality is not the same. For a fixed voice communication network, the required coding quality is not as high as that of a mobile voice communication network, so the receiving end can decide to decode according to the current network environment.
  • the number of layers If the current environment of the receiving end is a fixed network environment, the number of decoding layers selected is small, and the decoding rate when decoding the background noise coded stream is small; if the current environment of the receiving end is a mobile network environment, Then choose The number of decoding layers to be selected is large, and the decoding rate when decoding the background noise coded stream is large. In this way, under the condition that the background noise coded stream is layered, the receiving end can flexibly select the background noise decoding rate according to the current application environment.
  • the enhancement layer after the core layer is a supplement to the core layer, which can be effective. Enhance the coding effect of the core layer.
  • a frame used to describe a background noise coded stream in a speech coder is called a SID frame, and its coding rate is 1.8 kbit/s (one frame every 20 ms, 35 bits per frame, so the rate is 1.8 kbit/s). ), the specific frame content is shown in Figure 7.
  • the SID frame includes the filter parameter LSF and the energy parameter representing the excitation feature, which are encoded with 29 and 6 bits, respectively.
  • the SID frame is set as a core layer of the background noise coded code stream, and then on the basis of the core layer, some quantization bits that can improve the coding precision are added as enhancements.
  • Layer 1 for example, adding additional parameters that can characterize the background noise. Since the background noise coding method is a simple speech coding method, the characteristic parameters such as pitch delay in the speech encoder can be introduced into the background.
  • the quantization method of the specific pitch characteristic parameters can be quantized by the quantization method of the pitch delay characteristic parameters at the coding rate of 5.15 kbit/s, which needs to be encoded by 20 bits.
  • the encoder transmits the SID n frame including the core layer and the enhancement layer to the communication channel, and the communication channel can determine the number of layers to be received according to the current channel condition, and if the channel condition is good, the bandwidth is sufficient. , then the core layer and the enhancement layer are all received, as shown in Figure 5, otherwise only Receive the core layer, as shown in Figure 6.
  • the decoder can select the decoding layer according to the processing capability of the decoding end or the application environment, that is, the selection is 1.8 kbit. /s is still 2.8kbit/s for decoding.
  • the method is not limited to a speech codec, and includes an audio codec.
  • the number of layers of the actual background noise coded stream layer is not limited to two layers, and may have multiple layers.
  • an embodiment of the present invention provides a system for encoding code stream transmission of background noise, where the system includes an encoder 901 and a decoder 902, wherein an encoder 901 is configured to send background noise to a decoder at a coding rate.
  • the coded code stream includes a coding layer including a core layer and one or more enhancement layers, each layer includes a set of coding parameters, and each set of coding parameters corresponds to a coded bit number, and the coding rate is according to the
  • the coded code stream includes a number of coded bits corresponding to each layer, and the core layer includes a coding parameter that enables the background noise to be recovered by the receiver at the coding rate corresponding to the core layer, and the enhancement layer includes the enhancement layer.
  • a decoder 902 configured to receive an encoded code stream of background noise, determine a decoding layer, determine a decoding rate according to the decoding layer, and decode the encoded code stream at the rate;
  • the system further includes a communication channel 903 for transmitting an encoded code stream of background noise transmitted by the encoder to the decoder, and determining, during the transmission, the number of layers of the encoded code stream according to the current channel condition, according to the layer
  • the encoded code stream is received and the received encoded code stream is sent to the decoder.
  • an embodiment of the present invention further provides an encoder, which can be applied to a system for transmitting code stream of background noise, and is connected to a decoder, where the encoder includes a first processing unit 1001 for encoding
  • the rate sends a coded code stream of background noise to the decoder, where the coded code stream includes a coding layer including a core layer and one or more enhancement layers, each layer includes a set of coding parameters, and each set of coding parameters corresponds to a coded bit number.
  • the coding rate is determined according to the number of coding bits corresponding to each layer of the coded code stream, and the core layer includes coding parameters that enable the receiver to recover the background noise by itself when decoding the coding rate corresponding to the core layer.
  • the enhancement layer includes coding parameters that enhance the coding effect of the previous coding layer of the layer;
  • the first processing unit 1001 includes a first determining unit 10011 and a first sending unit 10012, where the first determining unit 10011 is configured to determine the encoding rate according to the number of encoding bits corresponding to each layer of the encoded code stream;
  • the unit 10012 is configured to send the coded code stream of the background noise to the decoder at the coding rate.
  • an embodiment of the present invention further provides a decoder, which can be applied to a system for encoding code stream transmission of the background noise, and is coupled to an encoder, where the decoder includes a second processing unit 1101 for receiving background noise.
  • the decoder includes a second processing unit 1101 for receiving background noise.
  • a coded stream determining a decoding layer according to a current processing capability or an application environment of the receiving end, determining a decoding rate according to the decoding layer, and decoding the encoded code stream at the rate
  • the second processing unit 1101 includes a receiving unit
  • the first determining unit 11011 is configured to receive the encoded code stream of the background noise
  • the second determining unit 11012 is configured to determine the decoding layer according to the current processing capability or the application environment of the receiving end. Determining a decoding rate according to the decoding layer; and decoding unit 11013, configured to decode the encoded code stream at the decoding rate.
  • the transmitting end when the transmitting end encodes the background noise, it does not need to consider the current specific situation of the communication channel, and the communication channel does not need to instruct the transmitting end to send the encoded code stream to which encoding rate, but directly
  • the encoded code stream is transmitted to the communication channel at an encoding rate, which simplifies the interaction of the communication channel with the transmitting end.
  • the coded code stream of the background noise includes a coding layer including a core layer and one or more enhancement layers, and the coding parameters included in the core layer enable the receiving end to recover the background noise when decoding at the coding rate corresponding to the core layer, and the enhancement layer
  • the included coding parameters serve to enhance the previous coding layer coding effect of the layer, which greatly improves the coding quality of the background noise.
  • the communication channel may determine the number of layers of the coded code stream according to the current channel condition and receive the coded code stream according to the layer number. If the channel condition is poor, the channel may select the received code. For fewer layers in the stream, if the channel conditions are good, the channel can choose to receive more layers in the code stream to make full use of the channel resources. Therefore, the communication channel can flexibly and freely select the number of layers of the received code stream according to its own situation, thereby flexibly controlling the coding quality.
  • the receiving end After receiving the background noise coded stream sent by the communication channel, the receiving end may also be based on the receiving end.
  • the specific processing capability or application environment determines the number of decoding layers, determines the decoding rate based on the number of layers, and awakens the encoded code stream at this rate. It can be seen that the receiving end can also flexibly select the decoding rate according to its own situation, thereby flexibly controlling the encoding quality, and it is possible for the receiving end to obtain high quality background noise.

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Description

一种背景噪声的编码码流传输的方法、 系统及装置 技术领域
本发明涉及语音通信领域， 尤其涉及一种背景噪声的编码码流传输的方 法、 系统及装置。 背景技术
在语音通信中， 对语音的处理主要由语音编解码器来完成， 最初的语音编 解码器都是定速率的， 即每一种语音编码器只有一个固定的速率。 这些传统 的定速率语音编码器从总体来讲， 较高速率的编码算法能较容易的保证编码 质量， 但占用通信信道资源较大； 较低速率的编码算法占用通信信道资源较 小， 但不太容易保证编码质量。
在语音通信中， 由于人发声并不是连续的， 大约有 70 %左右的空闲时间没 有讲话， 因此始终用同一个速率进行语音编解码显然是对通信信道资源的一 种极大的浪费。 基于这种考虑， 人们在语音编码器中引入了语音激活检测 ( Voice Activity Detector, VAD )技术， 这种技术能有效区分有话语音和无话 语音， 在有话语音时用正常的编码速率进行编码， 形成语音帧， 而无话语音 是一些背景噪声和静音， 因此可以用简单的编码方式进行处理， 形成静音描 述（Silence Descriptor, SID ) 帧。 通过这种变速率的编码方式， 有效的降低 了整个通话过程的平均编码速率， 节省了大量的通信信道资源， 而编码质量 也得到了较好的保证。
随着编码技术的发展， 同一个语音编码器中的语音编码速率并不仅仅局限 于针对语音的编码速率和针对背景噪声的编码速率两种。 这种编码器可以为 语音编码提供多种编码速率， 被称为变速率语音编码器。 由于变速率语音编 码器可以动态的调整编码速率， 使得语音通信系统在合成语音质量和系统容 量之间能够灵活的折中， 因此变速率语音编码器获得了快速的发展。 另外， 由于音频业务的快速发展， 语音编码器不仅仅只处理语音信号， 还会处理包 括音乐在内的各种音乐信号， 而这些音频信号所需要的编码速率也不太一样， 因此变速率语音编码器就显得更为重要。
编码器在选择编码速率时的依据主要有两个， 一个依据是语音信号本身的 特性， 即源控， 利用 VAD技术进行速率的选择就是一个源控的简单例子； 另 一个依据是通信信道的状况， 即信道控。 源控的一个典型例子是语音编码器 根据语音信号是浊音还是清音， 浊音是否稳定等信息对语音信号按照不同的 速率进行编码。 信道控的典型例子是语音编码器根据信道的状况要求编码器 对语音信号按照不同的速率进行编码， 若信道状况不好， 较为繁忙， 为节省 带宽， 就要求用较低的速率对语音信号进行编码， 否则若信道状况较好， 不 繁忙， 带宽充足， 就要求用较高的速率对语音信号进行编码， 以获得较高的 合成语音质量。
传统的变速率语音编码器中， 不同的编码速率通常对应不同的编码方式， 因此实际的编码码流也就大不相同。 而当前最新的语音编码器采用了一种全 新的编码速率形成方式， 这种编码速率形成方式的基础是编码器输出的语音 编码码流是以分层的方式封装成帧， 即编码码流由核心层和一个以上的增强 层组成， 每层为一组编码参数， 每組编码参数对应一编码比特数， 例如某一 语音的编码码流由核心层和 11个增强层组成， 如图 1所示：
层 1为占用编码比特数为 8kbit的核心层，层 2为占用编码比特数为 4kbit 的窄带增强层， 层 3〜12为占用编码比特数为 2kbit 的宽带增强层， 假如帧速 度为每秒传输一帧， 则该编码码流包含的层数分别为 1~12 时， 其对应的编 码速率分别为 8kbit/s、 12kbit/s、 14kbit/s、 16kbit/s、 18kbit/s、 20kbit/s、 22kbit/s、 24kbit/s、 26kbit/s、 28kbit/s、 30kbit/s和 32kbit/s。 基于这种速率形成方式， 其 各速率是向下兼容的， 以速率 12kbit/s为例， 其包含层 1和层 2 , 那么在接收 端除了能用 12kbit/s速率进行解码外，还可以丟掉层 2,保留层 1 , 即用 8kbit/s 的速率进行解码。
由于这种速率分级的特点， 信道不必指示编码器用哪一个速率进行编 码， 编码器会按 32kbitZs的最高速率对语音信号进行编码， 然后将这些编码码 流按层传送到通信信道， 信道依据当前的信道状况， 决定接收的层数。 若信 道状况较好， 信道容量较大， 不繁忙， 那么信道就会将这些编码码流全部接 收， 即接收所有的编码层； 若信道状况较差， 较为繁忙， 可用容量较小， 那 么信道的接收能力就受到了限制， 只会接收这些编码层中的前一部分， 而将 后一部分丢掉， 如丟掉层 7 ~ 12， 这样就剩下了层 6以下的部分， 也即接收层 1-6, 如图 2所示。
采用这样的速率控制方式， 语音编码器就不必考虑信道的具体情况而对每 种语音信号均按 32kbit/s速率进行编码， 并将这 32kbit/s的编码码流分层传送 到通信信道， 让信道根据实际的信道状况选择实际的传输速率， 因此这种速 率选择的信道控的方式简化了交互的过程， 使得信道对编码速率的控制更加 灵活。
在当前的语音编码器中， 在解码端， 背景噪声的合成原理与语音的合成原 理相同。 语音的合成原理是： 语音 可以看成是一个激励信号 e(n、激励一个 合成滤波器 νθ)所产生的输出，即 s(n) = e(n) * v(n) ,这就是语音产生的数学模型。 在合成背景噪声时用的也是这个模型， 所以背景噪声编码码流中所传输的描 述背景噪声和静音特性的特征参数内容与语音编码码流中的特征参数基本相 同， 为信号合成时的合成滤波器参数和激励参数。
在语音编码码流中， 合成滤波器参数主要为线语频率 （Line Spectium Frequence, LSF )量化参数， 而激励信号参数包括： 基音延迟参数、 基音增益 参数、 固定码本参数和固定码本增益参数。 不同的编码器， 这些参数的量化 比特数和量化形式有所不同； 相同的编码器， 如果其包含多个速率， 在不同 速率下， 由于描述信号特性的侧重点不同， 编码参数的量化比特数和量化形 式也有所不同。
与语音编码参数不同， 背景噪声编码参数描述的是背景噪声特性， 由于背 景噪声的激励信号可以认为是简单的噪声随机序列， 而这些序列在编解码端 均可以简单的用随机噪声产生模块产生， 然后用能量参数控制这些序列的幅 度， 就可产生最终的激励信号， 因此激励信号特征参数可以简单的用能量参 数来表示， 而不需要用其它的一些特征参数来进一步描述， 所以在背景噪声 编码码流中， 其激励参数为当前背景噪声帧的能量参数， 这与语音帧不同； 与语音帧相同的是，背景噪声编码码流中的合成滤波器参数也为线谱频率 LSF 量化参数， 只是具体的量化方法有所差别。 通过以上分析， 也可以认为背景 噪声编码码流本质上就是一种低速率的 "语音"编码码流。
各变速率语音编码器中针对语音或音频信号的速率较多， 而在处理背景 噪声时， 由于背景噪声编码帧描述的只是当前语音通信环境的背景噪声， 它 并不是通信的主体， 因此编码器在设计自己的背景噪声编码帧时都较为简单， 速率较低， 只是对背景噪声的一些简单信息进行了编码， 在解码端恢复出来 的背景噪声只是一个大致的编码端背景噪声的模拟， 并不十分精确。 因此也 可以理解为针对背景噪声编码的编码方式实际上就是一种简单的低速率语音 编码方式。
在变速率语音编码器中都有专门针对背景噪声编码的编码速率， 但是每 一个编码器中针对背景噪声进行编码的速率均只有一个， 在对背景噪声进行 编码时， 每一种编码器只用同一个固定的噪声编码速率也即同一种噪声编码 帧来对背景噪声进行编码。
在带宽资源充足时， 对通信双方来讲， 最重要的是通信的质量， 在通话 时， 需要用高质量的编码方式对话音进行编码， 在不通话时也需要用高质量 的编码方式对背景噪声进行编码。 前一点很容易满足， 因为变速率语音编码 器有众多的语音编码速率可供编码器选择， 但后一点无法满足， 因为不管带 宽多么充足， 对背景噪声进行编码时， 编码码流只对应一种编码速率， 这样 由于编码速率的限制， 虽然带宽允许， 通信信道只能直接接收发送端发来的 背景噪声编码码流， 并将该编码码流发送到接收端， 接收端也只能用该编码 速率进行解码， 通信信道和接收端无法对编码质量进行灵活选择和控制。 发明内容 本发明的实施例提供一种背景噪声的编码码流传输的方法、 系统及装置 , 用以解决现有技术中存在的接收端无法对编码质量进行灵活选择和控制的问 题。
本发明的实施例提供的一种背景噪声的编码码流传输的方法包括以下步 骤：
发送端以一编码速率向接收端发送背景噪声的编码码流，
所述编码码流包含的编码层包括核心层和一个以上的增强层， 每层包含 一组编码参数， 每组编码参数对应一编码比特数， 所述编码速率是根据所述 编码码流包含各层对应的编码比特数确定， 所述核心层包含使接收端以该核 心层对应的编码速率解码时能将该背景噪声独自恢复出来的编码参数， 所述 增强层包含增强该层以前的编码层编码效果的编码参数；
接收端接收到背景噪声的编码码流后， 确定解码层， 根据所述解码层确 定解码速率并以该速率对所述编码码流进行解码。
本发明的实施例提供的一种背景噪声的编码码流传输的系统， 该系统包 括：
编码器， 用于以一编码速率向解码器发送背景噪声的编码码流， 所述编 码码流包含的编码层包括核心层和一个以上的增强层， 每层包含一组编码参 数， 每组编码参数对应一编码比特数， 所述编码速率是根据所述编码码流包 含各层对应的编码比特数确定， 所述核心层包含使接收端以该核心层对应的 编码速率解码时能将该背景噪声独自恢复出来的编码参数， 所述增强层包含 增强该层以前的编码层编码效果的编码参数；
解码器， 用于接收背景噪声的编码码流， 根据接收端当前的处理能力或 应用环境确定解码层， 根据所述解码层确定解码速率并以该速率对所述编码 码流进行解码。
本发明的实施例提供的一种编码器， 与解码器相连， 该编码器包括： 第一处理单元， 用于以一编码速率向解码器发送背景噪声的编码码流， 所述编码码流包含的编码层包括核心层和一个以上的增强层， 每层包含 一组编码参数， 每组编码参数对应一编码比特数， 所述编码速率是根据所述 编码码流包含各层对应的编码比特数确定， 所述核心层包含使接收端以该核 心层对应的编码速率解码时能将该背景噪声独自恢复出来的编码参数， 所述 增强层包含增强该层以前的编码层编码效果的编码参数。
本发明的实施例提供的一种解码器， 与编码器相连， 该解码器包括： 第二处理单元， 用于接收背景噪声的编码码流， 根据接收端当前的处理 能力或应用环境确定解码层， 根据所述解码层确定解码速率并以该速率对所 述编码码流进行解码。
与现有技术相比， 本发明实施例中发送端以一编码速率将背景噪声的编 码码流发送到通信信道， 该编码码流包含核心层和一个以上的增强层， 该编 码码流到达通信信道后， 通信信道可根据信道状况对接收编码码流的层数进 行选择， 将接收的编码码流发送到接收端， 接收端还可根据当前的处理能力 或应用环境确定解码速率并以该速率对所述编码码流进行解码， 所以通信信 道和接收端可以对编码质量进行灵活选择和控制。 附图说明
图 1为现有技术中语音编码码流的结构示意图；
图 2 为现有技术中通信信道对接收语音编码码流的层数进行控制的示意 图；
图 3为本发明实施例中方法的流程示意图；
图 4为本发明实施例中背景噪声编码码流的结构示意图；
图 5 为本发明实施例中通信信道状况较好时接收背景噪声编码码流的示 意图；
图 6 为本发明实施例中通信信道状况较差时接收背景噪声编码码流的示 意图；
图 7为现有技术中背景噪声编码帧的比特分配图；
图 8为本发明实施例中背景噪声编码帧的比特分配图； 图 9为本发明实施例中系统的结构示意图；
图 10为本发明实施例中编码器的结构示意图;
图 11为本发明实施例中解码器的结构示意图。 具体实施方式
本发明实施例提供一种背景噪声的编码码流传输的处理方法， 该方法的 核心思想是： 发送端将包含核心层和一个以上增强层的背景噪声编码码流发 送到通信信道， 通信信道根据信道状况对接收编码码流的层数进行选择， 将 接收的编码码流发送到接收端 , 接收端根据该编码码流包含的层数确定解码 速率并以该速率对所述编码码流进行解码。
参见图 3，本发明提供的一种背景噪声的编码码流传输的方法的具体实施 流程包括以下步骤：
步骤 301 :
发送端对背景噪声编码时， 对编码码流采用分层的方式封装成帧， 封装 后的编码码流包含的编码层包括核心层和一个以上的增强层， 每层包含一组 编码参数， 其中核心层包含的编码参数能使接收端以该核心层对应的编码速 率解码时能将该背景噪声独自恢复出来， 增强层包含的编码参数能够增强该 层以前的编码层的编码效果。 并且， 每组编码参数对应一编码比特数， 核心 层对应的编码速率为该核心层对应的编码比特数乘以帧速度， 其中帧速度为 每秒发送的帧数。 发送端根据该编码码流中包含的各层对应的编码比特数确 定一编码速率， 确定方法为该编码码流中包含的各层对应的编码比特数之和 乘以帧速度， 然后发送端将背景噪声的编码码流以该编码速率发送到通信信 道。
步骤 302:
发送端发来的编码码流到达通信信道后， 通信信道根据当前的信道状况， 比如繁忙程度， 确定接收该编码码流的层数， 即决定接收到背景噪声编码码 流的哪一层， 并按照该层数接收该编码码流。 如果信道状况较差， 很繁忙， 带宽紧张， 就会丟弃增强层， 只接收核心层； 如果信道状况较好， 不繁忙， 带宽充足， 信道除了接收核心层外， 还可以接收增强层。 然后通信信道将接 收到的编码码流发送到接收端。
步骤 303:
接收端接收到通信信道发来的编码码流， 首先确定解码层数， 解码层数 确定后， 解码层也就相应确定， 例如， 若接收端确定的解码层数为 3 , 那么解 码层就为核心层、 增强层 1和增强层 2 , 若接收端确定的解码层数为 5 , 那么解 码层就为核心层、 增强层 1、 增强层 2、 增强层 3和增强层 4。 然后， 接收端根 据确定的解码层确定解码速率并以该速率对接收到的编码码流进行解码。
若接收端接收到的背景噪声编码码流只包含核心层， 那么解码层数为 1 , 解码层就为该核心层， 解码速率为该核心层对应的编码速率即该核心层对应 的编码比特数乘以帧速度得到的速率； 若接收端接收到的背景噪声编码码流 包含核心层和增强层， 那么接收端根据自身条件确定解码层数， 解码层也就 相应确定， 解码速率为该解码层对应的编码速率即该解码层对应的编码比特 数之和乘以帧速度得到的速率， 接收端的自身条件主要有两个：
(1)接收端的当前处理能力。 在对信号进行编解码时， 釆用的速率越高， 复杂度越大， 接收端所需要的处理能力越大， 因此在接收端， 接收端在对背 景噪声编码码流进行解码时也可以依据当前自身的处理能力决定解码层数。 若接收端当前的处理能力较小， 那么接收端选择的解码层数就较小， 相应的 解码速率也就较小； 若接收端当前的处理能力较强， 那么接收端选择的解码 层数就较大， 相应的解码速率也就较大。 这样接收端就可根据其当前的处理 能力， 灵活的进行解码速率的选择与控制。
(2)接收端的应用环境。 不同的应用环境， 所需的编码质量也不尽相同， 对固定语音通信网络来讲， 其所需的编码质量就不如移动语音通信网络的高， 因此接收端就可根据当前的网络环境决定解码层数。 若接收端当前的环境为 固定网络环境， 那么其选择的解码层数就较小， 对背景噪声编码码流进行解 码时的解码速率也就较小； 若接收端当前的环境为移动网络环境， 那么其选 择的解码层数就较大， 对背景噪声编码码流进行解码时的解码速率也就较大。 这样， 在背景噪声编码码流分层的条件下， 接收端就可根据当前的应用环境 灵活的选择背景噪声解码速率。
下面以一具体实施例对本发明提供的方法进行详细说明：
参见图 4， 某一背景噪声的编码码流包括核心层和增强层 1， 其中， 核心 层对应的编码比特数为 m(l)=30bit,增强层 1对应的编码比特数为 m(2)=5bit, 假设每秒 (s)传输 100 帧， 则发送端将该编码码流以 （30bit +5bit ) ΙΟΟ 帧 /s=3.5kbit/s的编码速率 2发送到通信信道。 接收端解码时采用核心层对应的 编码速率 1即 30bitx l00帧 /s=3kbit/s进行解码能独自将背景噪声恢复出来，核 心层之后的增强层是对核心层的一个补充， 其能有效的增强核心层的编码效 果。
现有技术中， 语音编码器中用来描述背景噪声编码码流的帧称为 SID帧， 其编码速率是 1.8kbit/s (每 20ms为一帧， 每帧 35bit, 所以速率是 1.8kbit/s ), 具体的帧内容如图 7所示。
通过图 7 , 可以发现， SID帧中包括滤波器参数 LSF和代表激励特征的能量 参数， 二者分别用 29和 6比特编码。 本实施例中， 可以以该 SID帧为基础， 将 该 SID帧设为背景噪声编码码流的核心层， 然后在其核心层的基础上， 再增加 一些能提高编码精度的量化比特， 作为增强层 1 , 比如增加额外的能表征背景 噪声特性的参数， 由于背景噪声编码方式就是一种简单的语音编码方式， 因 此， 可以将语音编码器中的特征参数如基音延迟这一特征参数引入到背景噪 声编码方式中来， 而具体的基音特征参数的量化方式可以采用 5.15kbit/s编码 速率下的基音延迟特征参数的量化方式来量化， 这需要用 20bit来编码。 这样 就形成了一个包括核心层和增强层 1的新的能更准确的描述背景噪声的 SID 帧， 速率为 2.8kbit/s , 可以称之为 SIDn帧， 如图 8所示。
在实际应用时，在背景噪声阶段，编码器将包括核心层和增强层的 SIDn帧 传送到通信信道， 通信信道就可根据当前的信道状况决定接收的层数， 若信 道状况良好， 带宽充足， 则将核心层和增强层全部接收， 如图 5所示， 否则只 接收核心层， 如图 6所示。
除了信道能够灵活的选择接收的层数， 在解码端， 若解码器接收到的是 SIDn帧， 则解码器就可以依据解码端的处理能力或应用环境来选择解码层， 即选择是以 1.8kbit/s还是 2.8kbit/s来进行解码。
由于越来越多的包括音乐在内的音频信号也需要在通信时编码传输， 因 此， 本方法并不局限于语音编解码器， 还包括音频编解码器。 另外实际的背 景噪声编码码流分层的层数不限于两层， 可以有多层。
参见图 9, 本发明实施例提供一种背景噪声的编码码流传输的系统， 该系 统包括编码器 901和解码器 902， 其中， 编码器 901， 用于以一编码速率向解 码器发送背景噪声的编码码流， 所述编码码流包含的编码层包括核心层和一 个以上的增强层， 每层包含一组编码参数， 每组编码参数对应一编码比特数， 所述编码速率是根据所述编码码流包含各层对应的编码比特数确定， 所述核 心层包含使接收端以该核心层对应的编码速率解码时能将该背景噪声独自恢 复出来的编码参数， 所述增强层包含增强该层以前的编码层编码效果的编码 参数； 解码器 902, 用于接收背景噪声的编码码流， 确定解码层， 根据所述解 码层确定解码速率并以该速率对所述编码码流进行解码；
该系统还进一步包括通信信道 903 ,用于传输编码器向解码器发送的背景 噪声的编码码流， 在传输过程中， 根据当前的信道状况确定接收所述编码码 流的层数， 按照该层数接收该编码码流， 并将接收的编码码流发送到解码器。
参见图 10, 本发明实施例还提供一种编码器， 可以应用于上述背景噪声 的编码码流传输的系统中， 与解码器相连， 该编码器包括第一处理单元 1001 , 用于以一编码速率向解码器发送背景噪声的编码码流， 所述编码码流包含的 编码层包括核心层和一个以上的增强层， 每层包含一组编码参数， 每组编码 参数对应一编码比特数， 所述编码速率是根据所述编码码流包含各层对应的 编码比特数确定， 所述核心层包含使接收端以该核心层对应的编码速率解码 时能将该背景噪声独自恢复出来的编码参数， 所述增强层包含增强该层以前 的编码层编码效果的编码参数； 第一处理单元 1001 包括第一判断单元 10011和第一发送单元 10012 , 其 中， 第一判断单元 10011 用于根据所述编码码流包含各层对应的编码比特数 确定所述编码速率； 第一发送单元 10012, 用于以所述编码速率向解码器发送 背景噪声的编码码流。
参见图 11， 本发明实施例还提供一种解码器， 可以应用于上述背景噪声 的编码码流传输的系统中， 与编码器相连， 该解码器包括第二处理单元 1101 , 用于接收背景噪声的编码码流， 根据接收端当前的处理能力或应用环境确定 解码层， 4艮据所述解码层确定解码速率并以该速率对所述编码码流进行解码； 第二处理单元 1101 包括接收单元 11011、 第二判断单元 11012和解码单 元 11013 , 其中， 接收单元 11011 , 用于接收背景噪声的编码码流； 第二判断 单元 11012, 用于根据接收端当前的处理能力或应用环境确定解码层， 根据所 述解码层确定解码速率； 解码单元 11013， 用于以所述解码速率对所述编码码 流进行解码。
本发明实施例提供的方法中， 发送端在对背景噪声进行编码时， 不必考 虑通信信道的当前具体状况， 通信信道也不必指示发送端按照哪种编码速率 向它发送编码码流， 而是直接将编码码流以一编码速率发送到通信信道， 这 样简化了通信信道与发送端的交互过程。
背景噪声的编码码流包含的编码层包括核心层和一个以上的增强层， 核 心层包含的编码参数能使接收端在以该核心层对应的编码速率进行解码时将 背景噪声恢复出来， 增强层包含的编码参数起到增强该层以前的编码层编码 效果的作用， 这样很大程度上提高了背景噪声的编码质量。
背景噪声的编码码流到达通信信道后， 通信信道可以根据当前的信道状 况决定接收该编码码流的层数并按照该层数接收该编码码流， 若信道状况较 差， 信道可以选择接收码流中的较少层， 若信道状况较好， 信道可以选择接 收码流中的较多层以充分利用信道资源。 所以， 通信信道可以根据自身情况 灵活自由地选择接收码流的层数， 从而对编码质量进行灵活控制。
接收端接收到通信信道发来的背景噪声编码码流后， 还可以根据接收端 当前的处理能力或应用环境等具体情况决定解码层数， 根据该层数确定解码 速率并以该速率对编码码流惊醒解码。 可见， 接收端也可以根据自身情况灵 活选择解码速率， 从而对编码质量进行灵活控制， 接收端能够得到高质量背 景噪声成为可能。
显然， 本领域的技术人员可以对本发明进行各种改动和变型而不脱离本 发明的精神和范围。 这样， 倘若本发明的这些修改和变型属于本发明权利要 求及其等同技术的范围之内， 则本发明也意图包含这些改动和变型在内。

Claims

权 利 要 求
1、 一种背景噪声的编码码流传输的方法， 其特征在于， 该方法包括以下 步骤：
发送端以一编码速率向接收端发送背景噪声的编码码流，
所述编码码流包含的编码层包括核心层和一个以上的增强层， 每层包含 一组编码参数， 每组编码参数对应一编码比特数， 所述编码速率是根据所述 编码码流包含各层对应的编码比特数确定， 所述核心层包含使接收端以该核 心层对应的编码速率解码时能将该背景噪声独自恢复出来的编码参数， 所述 增强层包含增强该层以前的编码层编码效果的编码参数；
接收端接收到背景噪声的编码码流后， 确定解码层， 根据所述解码层确 定解码速率并以该速率对所述编码码流进行解码。
2、 如权利要求 1所述的方法， 其特征在于， 所述发送端以一编码速率向 接收端发送背景噪声的编码码流包括：
发送端以一编码速率将背景噪声的编码码流发送给通信信道；
所述通信信道确定接收所述编码码流的层数， 并按照该层数接收该编码 码流；
所述通信信道将接收的编码码流发送到接收端。
3、 如权利要求 2所述的方法， 其特征在于， 所述通信信道根据当前的信 道状况确定接收所述编码码流的层数。
4、 如权利要求 2所述的方法， 其特征在于， 如果所述接收端接收到的编 码码流只包含核心层， 接收端将该核心层确定为解码层 。
5、 如权利要求 2所述的方法， 其特征在于， 如果所述接收端接收到的编 码码流包含核心层和增强层， 接收端根据自身当前的处理能力或应用环境确 定解码层。
6、 一种背景噪声的编码码流传输的系统， 其特征在于， 该系统包括： 编码器， 用于以一编码速率向解码器发送背景噪声的编码码流， 所述编 码码流包含的编码层包括核心层和一个以上的增强层， 每层包含一组编码参 数， 每组编码参数对应一编码比特数， 所述编码速率是根据所述编码码流包 含各层对应的编码比特数确定 , 所述核心层包含使接收端以该核心层对应的 编码速率解码时能将该背景噪声独自恢复出来的编码参数， 所述增强层包含 增强该层以前的编码层编码效果的编码参数；
解码器， 用于接收背景噪声的编码码流， 根据自身当前的处理能力或应 用环境确定解码层， 根据所述解码层确定解码速率并以该速率对所述编码码 流进行解码。
7、 如权利要求 6所述的系统， 其特征在于， 该系统进一步包括： 通信信道， 用于传输编码器向解码器发送的背景噪声的编码码流， 在传 输过程中， 根据当前的信道状况确定接收所述编码码流的层数， 按照该层数 接收该编码码流， 并将接收的编码码流发送到解码器。
8、 一种编码器， 与解码器相连， 其特征在于， 该编码器包括：
第一处理单元， 用于以一编码速率向解码器发送背景噪声的编码码流， 所述编码码流包含的编码层包括核心层和一个以上的增强层， 每层包含 一组编码参数， 每组编码参数对应一编码比特数， 所述编码速率是根据所述 编码码流包含各层对应的编码比特数确定， 所述核心层包含使接收端以该核 心层对应的编码速率解码时能将该背景噪声独自恢复出来的编码参数， 所述 增强层包含增强该层以前的编码层编码效果的编码参数。
9、 如权利要求 8所述的编码器， 其特征在于， 所述第一处理单元包括： 第一判断单元， 用于根据所述编码码流包含各层对应的编码比特数确定 所述编码速率；
第一发送单元， 用于以所述编码速率向解码器发送背景噪声的编码码流。
10、 一种解码器， 与编码器相连， 其特征在于， 该解码器包括： 第二处理单元， 用于接收背景噪声的编码码流， 根据自身当前的处理能 力或应用环境确定解码层， 根据所述解码层确定解码速率并以该速率对所述 编码码流进行解码。
11、如权利要求 10所述的解码器， 其特征在于， 所述第二处理单元包括: 接收单元， 用于接收背景噪声的编码码流；
第二判断单元， 用于根据自身当前的处理能力或应用环境确定解码层， 才艮据所述解码层确定解码速率；
解码单元， 用于以所述解码速率对所述编码码流进行解码。
PCT/CN2007/002680 2006-10-27 2007-09-11 Procédé, système et appareil pour transmettre le flux de code encodé du bruit de fond WO2008049311A1 (fr)

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