WO2006050657A1 - Method and device for adaptive multi-rate coding and transporting speech - Google Patents

Method and device for adaptive multi-rate coding and transporting speech Download PDF

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Publication number
WO2006050657A1
WO2006050657A1 PCT/CN2005/001803 CN2005001803W WO2006050657A1 WO 2006050657 A1 WO2006050657 A1 WO 2006050657A1 CN 2005001803 W CN2005001803 W CN 2005001803W WO 2006050657 A1 WO2006050657 A1 WO 2006050657A1
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Prior art keywords
frame
rate
amr
voice
adaptive multi
Prior art date
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PCT/CN2005/001803
Other languages
French (fr)
Chinese (zh)
Inventor
Wei Xiang
Original Assignee
Wei Xiang
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Filing date
Publication date
Priority claimed from CNB2004100680567A external-priority patent/CN1312946C/en
Priority claimed from CN2005100241263A external-priority patent/CN1829343B/en
Application filed by Wei Xiang filed Critical Wei Xiang
Publication of WO2006050657A1 publication Critical patent/WO2006050657A1/en

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Classifications

    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/04Speech 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 using predictive techniques
    • G10L19/16Vocoder architecture
    • G10L19/18Vocoders using multiple modes
    • G10L19/24Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding

Definitions

  • the present invention relates to a method for adaptive and multi-rate coding in a Universal Mobile Telecommunications System (UMTS) and for transmitting data generated by said coding, and related mobile stations, and in particular to improved adaptive multi-rate (AMR) coding Method of encoding and associated mobile station equipment, method of processing data output to an adaptive multi-rate (AMR) decoder, and associated mobile station equipment, access and control adaptive multi-rate during radio access ( AMR)
  • AMR adaptive multi-rate during radio access
  • AMR A method of encoding frames and a method of transmitting adaptive multi-rate (AMR) coded bits on a medium core network (CN) and a radio access network (RAN) interface.
  • variable rate speech coder performs the function of adaptive multi-rate (AMR) encoding
  • the central processor running the communication protocol stack software sets the mode of the variable rate speech coder, and is responsible for
  • TFC transport format combination
  • AMR adaptive multi-rate
  • the channel coder reads the adaptive multi-rate (AMR) encoded frame from the storage unit of the adaptive multi-rate (AMR) encoded frame generated by the variable rate speech coder; the channel decoder Complete channel decoding to generate an adaptive multi-rate (AMR) coded frame, and the variable rate voice decoder reads the adaptive multi-rate (AMR) coded frame from the memory unit of the channel adaptive decoder's placed adaptive multi-rate (AMR) coded frame for translation code.
  • AMR adaptive multi-rate
  • AMR adaptive multi-rate
  • TS Technical Specification
  • AMR Adaptive Multi-Rate
  • 3GPP TS 26.071 specifies that an adaptive multi-rate (AMR) encoder takes a 13-bit pulse code modulation (PCM) format voice signal as input, or converts other format voice signals to 13-bit precision.
  • a voice signal in a pulse code modulation (PCM) format is used as an input.
  • the encoder has a sampling rate of 8000 samples/second.
  • the encoder encodes a block of 160 samples of 13-bit precision pulse code modulation (PCM) format, resulting in encoding modes of 12.2, 10.2, 7.95, 7.40.
  • 3GPP TS 26.171 specifies that an adaptive multi-rate (AMR) encoder takes a 14-bit pulse code modulation (PCM) format voice signal as input, or converts other format voice signals into 14-bit precision pulses.
  • a voice signal in a coded modulation (PCM) format is used as an input.
  • the encoder has a sampling rate of 16,000 samples/second.
  • the encoder encodes a block of 320 14-bit pulse code modulation (PCM) format samples, producing encoding modes of 23.85, 23.05, 19.85, 18.25, 15.85. , a voice coded frame of 14.25, 12.65, 8.85, 6.6 kbps or a silence coded frame.
  • 3GPP TS 26.071 and TS 26.171 specify that an adaptive multi-rate (AMR) encoder can change the bit rate of a voice coded frame every 20 milliseconds according to a command to change the coding mode.
  • 3GPP TS 26.101 and TS 26.201 specify that in the content of each frame of the adaptive multi-rate (AMR) encoder output, the coding mode of the frame and the indication of the adaptive multi-rate (AMR) encoder should be indicated.
  • the coding mode and the request coding mode, and the coding mode of the frame, the indication coding mode, and the request coding mode are unique.
  • a core network exchanges signaling messages with the Radio Access Network (RAN) using the RANAP protocol, and transmits the attributes of the Radio Access Bearer (RAB) to the Radio Access Network (RAN), which includes adaptation.
  • AMR Multi-rate
  • a Radio Network Controller schedules a Radio Access Bearer Substream Combination Indicator (RFCI) according to the Radio Access Bearer (RAB) attributes, which correspond to the mode of an Adaptive Multi Rate (AMR) 20 msec voice frame.
  • RFCI Radio Access Bearer Substream Combination Indicator
  • RAB Radio Access Bearer
  • a radio network controller configures a number of dedicated channels (DCHs) that cooperate to work according to the RFCI, and one transport format (TF) in the DCH corresponds to one RFCI, so one RFCI corresponds to one or more transport format combination identifiers ( TFCI), the TFCI also corresponds to the mode of an adaptive multi-rate (AMR) 20 millisecond voice frame.
  • DCHs dedicated channels
  • TF transport format combination identifiers
  • AMR adaptive multi-rate
  • a radio network controller transmits a core network-radio access network interface initialization frame (Iu initialisation frame) to inform the core network of several RFCIs and associated radio access bearers (RABs) that it can provide to the pattern converter.
  • Iu initialisation frame a core network-radio access network interface initialization frame
  • RABs radio access bearers
  • a mobile station (UE) configures several DCHs that work cooperatively, and requires interaction of RRC signaling messages between a radio network controller (RNC) and a mobile station (UE) during configuration, and the mobile station (UE) can be based on a transport format.
  • RRC radio network controller
  • UE mobile station
  • TF informs the voice codec of the radio access bearer (RAB) sub-flow structure of the voice received and to be transmitted, the radio access bearer (RAB) sub-flow structure is Adaptive Multi-Rate (AMR) Class of bits for a 20 millisecond voice frame.
  • AMR Adaptive Multi-Rate
  • a piece of RFCI and its associated radio access bearer (RAB) substream size information obtained from the core network-radio access network (Iu) interface initialization frame (Iu initialisation frame) is sent to the code converter
  • the decoder is for configuration use.
  • the RPCI in the core network-radio access network user plane (Iu UP) frame is set to be the same as the adaptive multi-rate (AMR). a value corresponding to the mode of the 20 millisecond voice frame;
  • the core network-radio access network user plane (Iu UP) frame is decomposed into a plurality of DCHs working in cooperation, and the DCHs have different bit protection levels;
  • the TFCI of the radio frame is selected by the medium access layer MAC;
  • the mobile station (UE) receives the SDUs containing the voice frames from a number of DCHs that work together and restores them to an adaptive multi-rate (AMR) 20 millisecond voice frame, which can be used to indicate the structure of the voice frame to the voice codec.
  • AMR adaptive multi-rate
  • each 20 millisecond voice input is converted into a coded frame of a particular adaptive multi-rate (AMR) mode.
  • AMR adaptive multi-rate
  • the transmission data in the transmission channel is sent to the composite channel, and is subjected to operation steps such as CRC pasting, transport block concatenation/code block splitting, channel coding, etc., and becomes an input bit sequence of the rate matching module.
  • the output bit sequence that matches the number of bits of the radio frame of the physical channel by the rate matching output, when the rate matching puncturing has the maximum puncturing ratio limit, the transmission data that causes the number of puncturing to exceed the ratio limit will be Active discarding.
  • the maximum punch ratio is not limited for rate matching punches, those transmitted data with too large punch ratio cannot be correctly restored at the receiving end.
  • the physical channel rate equivalent to the number of physical frame radio frame bits can determine the validity of the transport channel format combination; the effective transport format combination concentrates those transmissions corresponding to the adaptive multi-rate (AMR) 20 millisecond voice frame mode.
  • the combination of channel formats determines the effective adaptive multi-rate (AMR) voice frame mode.
  • the Universal Mobile Telecommunications System introduces a combination of transport channel and transport format to schedule data from a logical channel to a transport channel during a periodic time interval. This time interval is called the transmission time interval (TTI) and the data is in accordance with the transport format (TF).
  • TTI transmission time interval
  • TF transport format
  • the transmission format is fixed during a given transmission time interval ( ⁇ ) interval, which also fixes the size, number, error correction coding mode, and rate matching parameter of the data block to be transmitted.
  • the transport format list of the transport channels used to construct the composite channel constitutes a Transport Format Combination (TFC).
  • the Radio Resource Control (RRC) unit configures a Transport Format Combination Set (TFCS) for the Media Intervention Control Unit MAC and Physical Layer.
  • RRC Radio Resource Control
  • TFC Transport Format Combination
  • a valid adaptive multi-rate (AMR) coded frame is three or two or one class of bits, each class occupying one transmission channel.
  • the prior art uses a fixed 20 millisecond transmission time interval ( ⁇ ) transport format combination to schedule a class of an adaptive multi-rate (AMR) frame to a transmission of the format using one of the combinations.
  • the method on the channel transmits an adaptive multi-rate (AMR) frame, and one frame outputted by the adaptive multi-rate (AMR) encoder every 20 milliseconds is carried by a transport block within a 20 millisecond transmission time interval (TTI), at each
  • TTI transmission time interval
  • the transmission format of the transmission channel corresponding to each class is selected according to the mode of the encoded frame output by the adaptive multi-rate (AMR) encoder, the format of the transmission channel of the voice data, and the format of the transmission channel of other data.
  • the transport format combination is formed, which of course is a combination of valid transport formats in the transport format combination set.
  • the radio link between the mobile station and the radio access network changes: this change is a temporary interruption of the radio link during hard handover; the soft handover is none Intermittent switching, but there is often a temporary decrease in the physical channel rate of the radio link during handover. It can be seen that this change in physical channel causes a loss of several adaptive multi-rate (AMR) 20 millisecond voice frames during handover.
  • AMR adaptive multi-rate
  • AMR adaptive multi-rate
  • the party transmitting the adaptive multi-rate (AMR) voice frame is sent at a fixed rate of one frame every 20 milliseconds. When a frame cannot be transmitted in a certain 20 milliseconds, the frame is discarded, and a voice frame is scheduled in the next 20 milliseconds.
  • An adaptive multi-rate (AMR) speech frame every 20 milliseconds from the encoding of an adaptive multi-rate (AMR) speech encoder speech. For each 20 millisecond voice, the air transmission time of its adaptive multi-rate (AMR) voice frame is fixed and unchangeable.
  • AMR adaptive multi-rate
  • UE mobile station
  • AMR adaptive multi-rate
  • UMTS Universal Mobile Telecommunications System
  • the technical problem to be solved by the present invention is to reduce the loss of an adaptive multi-rate (AMR) 20-millisecond speech frame during a voice call, although adaptive multi-rate (AMR) can avoid error frame loss by error concealment techniques. Negative effects, but this avoidance of error concealment techniques only works well for the loss of individual frames, because it uses the adjacent voice frames that have been received to construct the lost frames, so between the constructed lost frames and the actual lost frames. There is an error, and when the interval between lost frames is small or even continuous, the error of the subsequent lost frames becomes large.
  • AMR adaptive multi-rate
  • AMR adaptive multi-rate
  • a sudden drop in the physical channel rate can invalidate the original effective adaptive multi-rate (AMR) voice frame transmission format combination, causing adaptive multi-rate (AMR) voice frame loss.
  • AMR adaptive multi-rate
  • the adaptive multi-rate (AMR) voice coding mode generally requires a waiting time of 40 milliseconds or more, because the mobile station acquires the information of the physical channel rate change from the received radio frame to a time of 20 milliseconds of radio frame.
  • Changing the Adaptive Multi-Rate (AMR) voice coding mode takes at least 20 milliseconds to take effect; the second is to use the new Transport Format Combination (TFC), which is required when it is a Transport Format Combination (TFC) that needs to be configured. Signaling configuration process.
  • TFC Transport Format Combination
  • AMR transport adaptive multi-rate
  • TFC transport format combination
  • AMR transport adaptive multi-rate
  • the voice frame is lost, that is, the frame is stolen.
  • the proportion of non-voice services increases, and the complexity and number of signaling messages are greatly improved. More importantly, the composite channel mapped to the physical channel is used in the UMTS system to carry logical channels of all services.
  • the physical channel of the voice service is separated from the physical channel of the packet service, so that when the scheduling of the packet traffic channel preempts the physical channel, especially when the logical channel of the packet service has a higher priority than the voice logical channel, the voice
  • the number of times a channel is stolen is greatly increased, and a mechanism for reducing the negative impact of stealing frames is required.
  • the adaptive multi-rate (AMR) voice frame will be lost. If it is a soft handover and there is a temporary decrease in the physical link rate of the radio link, it may also cause Adapt to multi-rate (AMR) voice frame loss.
  • AMR adaptive multi-rate
  • the adaptive multi-rate (AMR) coded frame for each transport format combination selection as described in the background comes from an adaptive multi-rate (AMR) encoder, ie, adaptive multi-rate (AMR)
  • AMR adaptive multi-rate
  • the encoder refreshes the storage unit of the adaptive multi-rate (AMR) coded frame once every 20 milliseconds with the generated adaptive multi-rate (AMR) coded frame, and the channel encoder reads the adaptive from the memory unit every 20 milliseconds.
  • Multi-rate (AMR) encodes frames and encodes and multiplexes them.
  • the transport format combination selection is limited by those transport format combinations that are valid in the Transport Format Combination Set (TFCS) when performing the selection, often due to rapid channel changes, bursty signaling messages, and ".
  • TFCS Transport Format Combination Set
  • the present invention provides a technical solution for adaptive multi-rate (AMR) voice transmission control on a coding control, a transmission channel, and transmission and reception on a core network-radio access network (Iu) interface.
  • AMR adaptive multi-rate
  • AMR adaptive multi-rate
  • the scheme of encoding and transmitting in a mobile station of the present invention is a method of generating and transmitting an adaptive multi-rate (AMR) coded frame in a mobile station.
  • AMR adaptive multi-rate
  • an adaptive multi-rate (AMR) encoder passes The input voice signal is sampled to obtain a voice frame of 20 milliseconds in length. Each voice frame contains a sample sample of 20 millisecond voice, and encodes a 20 millisecond voice frame, which is generated in accordance with the Universal Mobile Telecommunications System (UMTS) standard.
  • UMTS Universal Mobile Telecommunications System
  • the wireless interface function unit is required to transmit voice data in the form of an adaptive multi-rate (AMR) coded frame, and the wireless interface function unit transmits the voice data according to the transmission format of each transport channel.
  • AMR adaptive multi-rate
  • An encoding command is issued to an adaptive multi-rate (AMR) encoder, the encoding command specifying a plurality of adaptive multi-rate (AMR) modes, the adaptive multi-rate (AMR) encoder according to the encoding command to a 20 millisecond voice
  • the effective voice frame coding sequence generated by frame coding is either a set of multiple adaptive multi-rate (AMR) voice coded frames or an adaptive multi-rate (AMR) silence coded frame, when the voice frame coding sequence is When multiple adaptive multi-rate (AMR) speech encoded frames, the mode of the adaptive multi-rate (AMR) encoded frame is consistent with the adaptive multi-rate (AMR) mode in the encoded command;
  • the process of transmitting an adaptive multi-rate (AMR) coded frame in a voice frame coding sequence to a transmission channel comprising selecting an adaptive multi-rate (AMR) coded frame in the voice frame coding sequence And selecting a transport format combination, the selected transport format combination, including the transmission format of all the bits of the selected adaptive multi-rate (AMR) coded frame, and using the transport format combination to select the adaptive multi-rate ( AMR)
  • the coded frame is scheduled onto the transport channel.
  • the remaining adaptive multi-rate (AMR) coded frames in the voice frame coding sequence are discarded after transmission of an adaptive multi-rate (AMR) coded frame is selected.
  • the above-mentioned wireless interface functional unit in the scheme of coding control and transmission in the mobile station is a functional unit that provides an air interface for the user data of the mobile station, and its input is user data such as voice, packet and signaling, and the output is a radio frame. It is compliant with communication protocols for Radio Resource Control (RRC), Radio Link Control (RLC), Media Access Control Unit (MAC), and Physical Layer (PHY).
  • RRC Radio Resource Control
  • RLC Radio Link Control
  • MAC Media Access Control Unit
  • PHY Physical Layer
  • TFC transport format combination
  • TFC transport format combination
  • AMR Transport Format Combination
  • TFC Transport Format Combination
  • the first in first out The (FIFO) buffer replaces the voice frame coding sequence that the original memory unit accepts from the adaptive multi-rate (AMR) encoder output.
  • the present invention provides a mobile station in a universal mobile communication system, the mobile station's adaptive multi-rate (AMR)
  • the encoder encodes a 20 millisecond length voice frame encoding to produce an output of the voice frame encoding sequence
  • the mobile station includes a first in first out (FIFO) buffer coupled to the output of the adaptive multi-rate (AMR) encoder, the first in first out The (FIFO) buffer includes - a data input interface for reading a sequence of voice frame codes generated by adaptive multi-rate (AMR) encoder encoding,
  • a data output interface for reading a stored voice frame code sequence from the first in first out (FIFO) buffer, a memory state interface for outputting a voice frame code sequence stored in the first in first out (FIFO) buffer The number and type of voice frame encoding sequence in it.
  • the storage of the first-in-first-out (FIFO) buffer is limited to a voice frame coding sequence
  • the voice frame coding sequence is subsequently written. cover.
  • more than two voice frame coding sequences can be stored in the above-mentioned first-in, first-out (FIFO) buffer, so that it is easy to read after 20 milliseconds of failure to read, gp, using the following A method of generating and transmitting adaptive multi-rate (AMR) coded frames in a discontinuous manner in a mobile station.
  • AMR adaptive multi-rate
  • the mobile station reads the type of the voice frame coding sequence to be processed from the storage state interface of the first in first out (FIFO) buffer, and when the voice frame coding sequence to be processed is a valid voice frame coding sequence to be transmitted, the voice is The frame coding sequence selects a transport format combination, and transmits an adaptive multi-rate (AMR) coded frame in the voice frame coding sequence in combination with the selected transmission format.
  • AMR adaptive multi-rate
  • the transport format (TF) configured according to the radio interface protocol functional unit of the mobile station can determine the class bits of the adaptive multi-rate (AMR) mode encoded frame corresponding thereto, and then determine according to the configured transport format combination (TFC). It corresponds to an adaptive multi-rate (AMR) mode so that an optional range of modes in the encoded command can be obtained from the configured Transport Format Combination Set (TFCS).
  • AMR adaptive multi-rate
  • TFC transport format combination
  • TFCS Transport Format Combination Set
  • a scheme for determining encoding control and transmission of a mode in an encoding command according to a radio interface protocol in a mobile station is: a method of generating and transmitting an adaptive multi-rate (AMR) encoded frame in a mobile station as described above, each of said encoding commands
  • the adaptive multi-rate (AMR) mode corresponds to a voice transmission format combination of the mobile station's transport format combination set, which includes an adaptive multi-rate (AMR) mode coded frame for all types of bit transmissions.
  • the format of the transport format combination is: a method of generating and transmitting an adaptive multi-rate (AMR) encoded frame in a mobile station as described above, each of said encoding commands
  • the adaptive multi-rate (AMR) mode corresponds to a voice transmission format combination of the mobile station's transport format combination set, which includes an adaptive multi-rate (AMR) mode coded frame for all types of bit transmissions.
  • the format of the transport format combination is: a method of generating and transmitting an adaptive multi-
  • the mobile station can match the mode of the frame output by the adaptive multi-rate (AMR) encoder with the changed transport format combination (TFC), and the mobile station wireless interface protocol functional unit can wirelessly The changes on the link are reflected to the adaptive multi-rate (AMR) encoder.
  • the wireless interface protocol function unit can detect such a fast change state and can request The encoder adapts to this fast change in a manner that outputs multi-mode adaptive multi-rate (AMR) encoded frames.
  • AMR Adaptive multi-rate
  • the coding and transmission scheme on the network side of the Universal Mobile Telecommunications System is similar to that in the mobile station, and the present invention provides an adaptive multi-speed generation and transmission on the network side of the Universal Mobile Telecommunications System (UMTS).
  • Rate (AMR) method of encoding frames during a voice call, a pattern converter in the core network converts the input voice encoded signal into an adaptive multi-rate (AMR) encoded frame, said adaptive multi-rate (AMR)
  • the coded frame is placed on the Iu user plane frame and sent to the Radio Network Controller (RNC), and the Radio Network Controller (RC) transmits voice data on the transport channel, characterized by - transmitting an encoding command to the pattern converter,
  • the encoding command specifies a plurality of adaptive multi-rate (AMR) modes, and the pattern converter encodes the input voice encoded signal according to the encoding command, and generates a valid voice frame encoding sequence of 20 milliseconds length voice.
  • AMR adaptive multi-rate
  • AMR adaptive multi-rate
  • AMR adaptive multi-rate
  • a Radio Network Controller the process of scheduling an adaptive multi-rate (AMR) coded frame in a voice frame coding sequence to a transmission channel, including selecting an adaptive multi-rate (AMR) in the sequence of voice frame codes Encoding the frame and selecting a combination of transmission formats, and selecting the combination of the transmission formats, including the transmission format of all the bits of the selected adaptive multi-rate (AMR) coded frame, and using the combination of the transmission formats to select the adaptive Multi-rate (AMR) coded frames are scheduled onto the transport channel.
  • the remaining adaptive multi-rate (AMR) coded frames in the voice frame coding sequence are discarded after transmission of an adaptive multi-rate (AMR) coded frame is selected.
  • the scheme for determining the encoding generation and transmission of the mode in the encoding command by the radio network controller (RNC) on the network side of the Universal Mobile Telecommunications System (UMTS) is generated and transmitted in the network side of the Universal Mobile Telecommunications System (UMTS) as described above.
  • a method for adapting a multi-rate (AMR) coded frame the radio network controller (RC) transmits a core network-radio access network interface (Iu) user plane rate control frame to the core network, and a core network-radio access network interface (Iu)
  • the radio frequency identification indicator (RFCI n indicator) field of the plurality of radio access bearer substream combination indications in the user plane rate control frame is corresponding to a plurality of radio access bearer substream combination indications (RFCI), and the radio access bearers are
  • the stream combination indication (RPCI) corresponds to an adaptive multi-rate (AMR) mode, and after the core network receives the core network-radio access network interface (Iu) user plane rate control frame, sends the signal to the pattern converter.
  • Encoding command, the multiple adaptive multi-rate (AMR) modes specified in the coding command are the multiple adaptations corresponding to the radio access bearer substream combination indication (RFCI) Rate (AMR) voice mode.
  • the present invention proposes a method for delay transmission of adaptive multi-rate (AMR) coded frames during transmission of voice channels or physical channels to stop transmitting voice data, and corresponding mobile station adaptive multi-rate (AMR) delay translation.
  • AMR adaptive multi-rate
  • the method of the code Delayed transmission is combined with the above-mentioned method of generating and transmitting a voice frame coding sequence, because the channel when transmitting again after delay may be different from the channel originally intended to be transmitted, and the original adaptive multi-speed cannot be used.
  • the rate (AMR) voice mode must look for a suitable adaptive multi-rate (AMR) mode.
  • the characteristics of real-time voice are:
  • the end-to-end delay is small, and the maximum allowable end-to-end delay is determined according to the person's perception of the audio session.
  • the subjective evaluation of the end-to-end delay is acceptable between 200-300 milliseconds. Therefore, as long as the method of the present invention controls the end-to-end delay to less than 200 milliseconds, it does not cause significant degradation in service quality in terms of delay.
  • AMR mobile station delay adaptive multi-rate
  • the mobile station adaptive multi-rate (AMR) decoder delay decoding scheme proposed by the present invention is: a method for a mobile station to process an adaptive multi-rate (AMR) coded frame, the mobile station transmitting voice An adaptive multi-rate (AMR) coded frame is received on the transport channel, and the adaptive multi-rate (AMR) decoder performs the adaptive multi-rate (AMR) coded frame received by the mobile station in a first to last order every 20 milliseconds.
  • a decoded output of an adaptive multi-rate (AMR) coded frame characterized by - setting up a buffer area for placing an adaptive multi-rate (AMR) coded frame for a voice call and a lower limit value of the buffer area, from the mobile station.
  • AMR adaptive multi-rate
  • AMR adaptive multi-rate
  • the received adaptive multi-rate (AMR) voice coded frame is stored in the buffer area, and the adaptive multi-rate is in the buffer area ( AMR)
  • the adaptive multi-rate (AMR) decoder begins decoding the adaptive multi-rate (AMR) coded frame received by the mobile station.
  • SPEECH voice
  • AMR adaptive multi-rate
  • AMR cache adaptive multi-rate
  • the subsequent transmission involves selecting a mode of frame transmission from a plurality of mode adaptive multi-rate (AMR) voice coded frames in the sequence, and transmitting the adaptive multi-rate (AMR) voice coded frame
  • AMR adaptive multi-rate
  • the Radio Network Controller (RNC) can also be a mobile station.
  • the adaptive multi-rate (AMR) coded frame uses the transparent mode RLC layer mode, and does not add a sequence number to the protocol data unit carrying the adaptive multi-rate (AMR) coded frame, and the receiver follows the received adaptive multi-rate (AMR) coded frame.
  • the chronological ordering if there is no adaptive multi-rate (AMR) coded frame available for the receiver in two radio frames every 20 milliseconds, the receiver inserts a received lost frame, and the type of the received lost frame is 3GPP TS26.
  • the reception type (RX_TYPE) in the 101 table of Table 101 (Table lc) is of the type of no data (NO-DATA).
  • the adaptive multi-rate (AMR) voice coded frame transmission is performed every 20 milliseconds in an uneven manner in the method of the present invention, it is necessary to add the received loss frames added by the error caused by the uneven transmission from the buffer area. Deleting, eliminating the negative effects caused by unevenly transmitting voice frames, such a method of utilizing the buffer storage space to accommodate temporary pauses in voice transmission such that the time of the adaptive multi-rate (AMR) voice coded frames at the time of encoding completion The distance is kept to the maximum extent.
  • the mobile station has two modes of operation-one-speech (SPEECH) mode and comfort noise (COMFORT-NOISE) mode.
  • SPEECH comfort noise
  • COMFORT-NOISE comfort noise
  • the sender intermittently transmits a silence frame
  • the receiver receives the lost frame that is actively added. Effective.
  • SPEECH voice
  • AMR multi-rate
  • AMR peer multi-rate
  • Received lost frames that do not have a corresponding multi-rate (AMR) voice-coded frame are redundant and do not compensate for the loss of multi-rate (AMR) voice-coded frames. If they cannot be deleted by the delete operation of the uneven transmission mode, the voice will be caused. The delay and quality are degraded, so there is a certain limit on the number of actively added received lost frames:
  • a buffer adaptive multi-rate (AMR) coded frame of the mobile station and a receiving method for adding and deleting a received lost frame, setting a limit number of the received lost frame actively added in the voice (SPEECH) mode of the buffer area, When the number of received lost frames actively added in the voice zone (SPEECH) mode reaches the limit number, the received loss frame is no longer added in the voice (SPEECH) mode.
  • AMR buffer adaptive multi-rate
  • the above limitation is imposed on the number of actively added received lost frames because: the number of delayed multi-rate (AMR) coded frames that can be accommodated on the path from the transmitting mobile station to the receiving mobile station's RNC is limited, Therefore, the number of received loss frames actively added in the SPEECH mode corresponding thereto is also limited, and the number limit value should be included to reflect this limitation.
  • AMR delayed multi-rate
  • the mobile station proposed by the present invention that is, a mobile station in a universal mobile communication system, the mobile station including a first in first out connection connected to an input of an adaptive multi-rate (AMR) decoder (FIFO) cache, this first in first out (FIFO) cache includes:
  • AMR adaptive multi-rate
  • a data output interface for reading the stored adaptive multi-rate (AMR) encoded frame from the first in first out (FIFO) buffer
  • a storage status interface for outputting the length of the adaptive multi-rate (AMR) encoded frame queue stored in the first in first out (FIFO) buffer and the type of the encoded frame
  • control unit configured to delete the adaptive multi-speed stored in the first in first out (FIFO) buffer according to the delete instruction Rate-of-Availability (AMR) encoded non-data (NO-DATA) type of adaptive multi-rate (AMR) coded frames in a frame queue.
  • AMR Rate-of-Availability
  • NO-DATA non-data
  • AMR adaptive multi-rate
  • Radio Network Controller For a Radio Network Controller (RNC) that transmits an Adaptive Multi-Rate (AMR) coded frame, it can use the receiver's mobile station's receive buffer lower-limit value to control its adaptation in the transmit buffer when switching.
  • the number of multi-rate (AMR) coding sequences therefore, a mechanism is needed to transmit the receiving buffer lower limit value of the receiving mobile station, and the present invention proposes that the mobile station sends a message containing the buffer lower limit value to the network side.
  • the lower limit value is transmitted to the core network and the radio network controller (RNC), and the network side delay mode is sent to control the delay, and the mobile station can also limit the buffer area of the scheme in the intermittent manner. This is used as a reference when setting the value.
  • AMR adaptive multi-rate
  • the technical solution for the intermittent transmission of the mobile station is: according to the above-mentioned method for generating and transmitting an adaptive multi-rate (AMR) coded frame in the mobile station, during the time when the mobile station pauses to transmit the radio frame carrying the voice data from the adaptive multi-rate (AMR) encoder's voice frame coding sequence is buffered, and after the pause is completed, more than one every 20 milliseconds is scheduled during the transmission of the adaptive multi-rate (AMR) coded frame in the voice frame coding sequence to the transmission channel.
  • Adaptive Multi-Rate (AMR) A method of encoding a frame to a transmission channel, processing some or all of the voice frame coding sequences that are buffered.
  • the prior art uses a fixed 20 millisecond transmission time interval (TTI) transport format combination to schedule a class of adaptive multi-rate (AMR) frames to a transmission of the format using one of the combinations.
  • TTI transmission time interval
  • the method on the channel transmits an adaptive multi-rate (AMR) frame, and the present invention will no longer adopt a fixed transmission time interval (TTI), but instead selects a transmission sequence of the to-be-processed voice frame to be transmitted.
  • a time interval ( ⁇ ) variable transport format combination such as a transport format combination with a transmission time interval (TTI) of 10 milliseconds to transmit an adaptive multi-rate (AMR) frame of one of the voice frame coding sequences.
  • the value of the transmission time interval (TTI) can be determined according to the number of voice frame coding sequences stored in the first in first out (FIFO), one method is
  • a transmission format combination having a transmission time interval (TTI) of 10 milliseconds is preferred, and the present invention suggests that the specified number is one.
  • a mobile station pauses to transmit a radio frame, which belongs to the category of a radio frame that suspends transmission of voice data.
  • Cell handover is the most common, and other: dynamic channel adjustment, that is, the mobile station switches from one physical channel to Another physical channel.
  • the mobile station does not suspend the transmission of the radio frame during the stealing of the frame, it belongs to the category of the radio frame in which the transmission of the voice data is suspended.
  • the voice frame coding is performed.
  • Adaptive Multi-Rate (AMR) in the sequence to encode a frame to the transmission channel, scheduling one every 20 milliseconds More than one (excluding one) adaptive multi-rate (AMR) coded frame to transport channel, processing part or all of the voice frame coding sequence being buffered.
  • An adaptive multi-rate (AMR) coded frame that cannot be transmitted during a cell handover.
  • the buffered adaptive multi-rate (AMR) coded frame is transmitted at a rate of more than one every 20 milliseconds. There is no pause in voice delivery during this period. This is the advantage of this method, which overcomes the discomfort that the voice frame loss caused by the handover brings to the listener.
  • the technical solution for interrupting the transmission of the buffered voice data is: according to the above method of generating and transmitting an adaptive multi-rate (AMR) coded frame in the mobile station, prioritizing the scheduling of other logical channels higher than the priority of the voice logical channel to During the pause of voice data transmission on the transport channel, the voice frame coding sequence from the adaptive multi-rate (AMR) encoder is buffered, and the adaptive multi-rate (AMR) in the transmitted voice frame coding sequence is recovered after the voice data transmission is restored.
  • AMR adaptive multi-rate
  • one or more (excluding one) adaptive multi-rate (AMR) coded frames are transmitted to the transmission channel every 20 milliseconds, and part or all of the voice frame coding sequences that are buffered are processed.
  • AMR adaptive multi-rate
  • the buffer is interrupted by the scheduled high-priority logical channel to intercept the adaptive multi-rate (AMR) encoded frame on the voice logical channel.
  • AMR adaptive multi-rate
  • the cache is adaptive. Rate (AMR) coded frames are transmitted at more than one rate every 20 milliseconds, which solves the problem of such framed data being stolen by the logical channel priority. This is the benefit of this method.
  • AMR adaptive multi-rate
  • AMR adaptive multi-rate
  • the radio network controller (RNC) on the network side detects whether the received adaptive multi-rate (AMR) coded frame is a delayed frame.
  • One method of detection is to check the adaptive multi-rate (AMR) received every 20 milliseconds.
  • the coded frame once it is found that more than one frame is received within 20 milliseconds or a frame is received within 10 milliseconds, indicating that the delayed frame is received; another method of detecting is to send signaling to the radio access network.
  • a message indicating the starting frame number and the number of frames of the transmitted radio frame of a number of delayed frames.
  • a method for a Radio Network Controller (RNC) to transmit an adaptive multi-rate (AMR) coded frame with a delay flag to the core network is: placing a delay flag on the bearer Should be multi-rate (AMR) encoded frame core network - radio access network interface (lu) user plane frame extension field (spare extension).
  • RNC Radio Network Controller
  • AMR adaptive multi-rate
  • a discontinuous transmission scheme on the network side of the Universal Mobile Telecommunications System is: a method of generating and transmitting an adaptive multi-rate (AMR) coded frame on the network side of the Universal Mobile Telecommunications System (UMTS) as described above.
  • AMR adaptive multi-rate
  • the received voice frame coding sequence carried by the lu user plane frame sent by the core network is buffered in the buffer area, and after the pause is finished, the adaptive multi-rate (AMR) coding in the buffered voice frame coding sequence is scheduled.
  • AMR adaptive multi-rate
  • one or more (excluding one) adaptive multi-rate (AMR) coded frames are transmitted to the transmission channel every 20 milliseconds.
  • the technical solution for transmitting on the network side of the Universal Mobile Telecommunications System (UMTS) at the time of cell handover is: according to the discontinuous transmission method on the network side of the Universal Mobile Telecommunications System (UMTS), the paused radio frame transmission period is determined by the radio network controller Inter-cell handover period within (RNC).
  • RNC Radio Network Controller
  • the technical solution of the Universal Mobile Telecommunications System (UMTS) network side transmission when switching between Radio Network Controllers (RNCs) is: According to the discontinuous transmission method on the network side of the Universal Mobile Telecommunications System (UMTS), the suspended radio frame transmission The period is during the switching between radio network controllers (RNCs).
  • RNCs Radio Network Controllers
  • the technical solution for transmitting on the universal mobile communication system (UMTS) network side when the voice data is interrupted by the high priority logical channel is: according to the intermittent transmission method on the network side of the Universal Mobile Telecommunications System (UMTS), the pause wireless
  • the frame transmission period is caused by the failure to transmit those core network-radio access network interface (lu) user plane frames caused by the suspension of voice data transmission when the other logical channels are preferentially scheduled to be higher than the voice logical channel priority to the transmission channel.
  • the period of the adaptive multi-rate (AMR) encoded frame in the voice frame coding sequence is caused by the failure to transmit those core network-radio access network interface (lu) user plane frames caused by the suspension of voice data transmission when the other logical channels are preferentially scheduled to be higher than the voice logical channel priority to the transmission channel.
  • AMR delayed adaptive multi-rate
  • a delayed adaptive multi-rate (AMR) coded frame is generated, causing the radio network controller (RNC) transmitting the mobile station to exceed one voice frame coding sequence every 20 milliseconds.
  • the rate of transmission of voice data to the core network the core network will also transmit to the radio network controller (RNC) of the receiving mobile station at a rate of more than one voice frame coding sequence every 20 milliseconds, a universal mobile communication system that processes such frames
  • the transmission scheme of the UMTS) network side is: a method for generating and transmitting an adaptive multi-rate (AMR) coded frame on the network side of the Universal Mobile Telecommunications System (UMTS) as described above, and in a universal mobile communication system (UMTS) a method for determining, by a radio network controller (RNC), a code generation and transmission of a mode in an encoding command, the lu user plane frame transmitted by the core network to a radio network controller (RNC) carrying a voice frame coding sequence,
  • RNC radio network controller
  • AMR adaptive multi-rate
  • AMR adaptive multi-rate
  • TTI transmission time interval
  • AMR adaptive multi-rate
  • the voice data is interrupted by the high priority logical channel.
  • RNC Radio Network Controller
  • the delay of the voice is required to be controlled, so the buffer is stored in the radio network controller (RNC).
  • the number of voice frame coding sequences in the buffer area is limited. When the number of buffered voice frame coding sequences exceeds the lower limit of the buffer area of the destination mobile station, the buffered voice frame coding sequence is discarded.
  • the present invention enables an adaptive multi-rate (AMR) mode selection mechanism in the TFC selection phase, which reduces the loss of voice frames caused by the single adaptive multi-rate (AMR) mode and the TF mismatch in the TFC selection phase in the prior art.
  • the output of the adaptive multi-rate (AMR) encoder is multi-mode, and this multi-mode has a combination of transport channel formats to match.
  • the output of the multi-mode adaptive multi-rate (AMR) encoder in the method of the present invention can be used for abrupt changes in the effective transport channel format combination (TFC) caused by a sudden drop in the physical channel rate or scheduling of burst high priority logical channels.
  • TFC effective transport channel format combination
  • TFC effective transmission channel format combination
  • the adaptive multi-rate (AMR) encoder outputs a multi-mode adaptive multi-rate (AMR) coded frame in a first-in, first-out (FIFO) buffer, and the multi-mode has a transport channel format combination thereof.
  • Match Abrupt changes in the effective transport channel format combination (TFC) caused by a sudden drop in the physical channel rate, handover, or scheduling of bursty high priority logical channels, ie, the transport channel format combination that can be used in the Transport Channel Format Combination Set (TFCS) This includes the Transport Channel Format Combination (TFC) used by the mobile station before this.
  • TFC Transport Channel Format Combination
  • the output of the multi-mode adaptive multi-rate (AMR) encoder in the method of the present invention can be used to match the abrupt changes in the effective transport channel format combination (TFC), ie, using a new available transport channel format combination (TFC) ) to schedule the adaptive multi-rate (AMR) transmitted by the previous mobile station in the sequence of voice frames to be processed in the first-in, first-out (FIFO) buffer Adaptive multi-rate (AMR) coded frames of different modes of mode, thereby reducing frame loss of voice frames caused by sudden drops in physical channel rates, handover or scheduling of bursty high priority logical channels, compared to the prior art
  • the present invention enables the selection of the best transport channel format combination (TFC) for the adaptive multi-rate (AMR) mode of all Universal Mobile Telecommunications System (UMTS) and the radio resources of the mobile station in a shorter time.
  • the buffering mechanism of the present invention overcomes the limitation that an adaptive multi-rate (AMR) voice frame transmission or discarding must be completed every 20 milliseconds in the prior art, because delayed transmission can be achieved using the method of the present invention, thus making handover
  • the coded frames output by the adaptive multi-rate (AMR) encoder during the scheduling of the burst high-priority logical channel preemption can still be transmitted, and the prior art will discard these switched and scheduled burst high-priority logical channel preemptions.
  • An adaptive multi-rate (AMR) coded frame of a radio resource that is, an adaptive multi-rate (AMR) that occurs in a storage unit during handover and scheduling of burst high-priority logical channels to preempt wireless resources in the prior art.
  • the phenomenon that the encoded frame is refreshed without being read.
  • the voice frames affected by the scheduled burst high priority logical channels in the method of the present invention may be transmitted in a different mode than the voice frames adjacent thereto before being delayed, or delayed after a certain time. Sending does not cause discarding of voice frames. This is reflected in the fact that it can adjust the timing of the first-in, first-out (FIFO) cache read adaptive multi-rate (AMR) coded frame.
  • FIFO first-in, first-out
  • AMR adaptive multi-rate
  • 10 can be used.
  • the transmission format of the millisecond transmission time interval (TTI) combines the adaptive multi-rate (AMR) encoded frames transmitted in the first in first out (FIFO) buffer to reduce the time interval between the writing and reading.
  • the method for generating and transmitting adaptive multi-rate (AMR) coding maximizes the use of the bit rate provided by the physical channel to transmit voice data. This utilization is reflected in the 20 millisecond voice input. Select within a certain mode range and within a certain time range, and the adaptation of this choice to the physical channel.
  • AMR adaptive multi-rate
  • Figure 2 is a process diagram of an encoding module having an adaptive multi-rate (AMR) mode number of 2 in the encoding command of Figure 1. .
  • AMR adaptive multi-rate
  • Figure 3 is a process diagram of an encoding module having an adaptive multi-rate (AMR) mode number of 3 in the encoding command of Figure 1.
  • AMR adaptive multi-rate
  • FIG. 4 is a block diagram of an embodiment of a mobile station generating and transmitting adaptive multi-rate (AMR) coded frames in a buffered manner.
  • AMR adaptive multi-rate
  • Figure 5 is an interface block diagram of a first-in, first-out (FIFO) buffer for an adaptive multi-rate (AMR) encoder.
  • Figure 6 to Figure 10 are diagrams showing the processing of the voice frame coding sequence in the first in first out (FIFO) buffer during the period from 0 to 80 milliseconds shown in Table 1 by the mobile station;
  • FIFO first in first out
  • FIG. 6 is a schematic diagram of a zero millisecond first in first out (FIFO) buffer output adaptive multi-rate (AMR) coded frame
  • FIG. 7 is a schematic diagram of outputting an adaptive multi-rate (AMR) encoded frame in a 0 millisecond first in first out (FIFO) buffer
  • Figure 8 is a diagram showing an adaptive multi-rate (AMR) encoder outputting a sequence of voice frame codes to a first in first out (FIFO) buffer at 35 milliseconds;
  • AMR adaptive multi-rate
  • Figure 9 is an illustration of a 40 millisecond first-in, first-out (FIFO) buffer output adaptive multi-rate (AMR) coded frame;
  • Figure 10 is a schematic illustration of a 40 millisecond first-in, first-out (FIFO) buffer output adaptive multi-rate (AMR) coded frame.
  • FIFO first-in, first-out
  • AMR adaptive multi-rate
  • 11 is a block diagram of an embodiment of delay decoding by a mobile station.
  • FIG. 12 is an interface block diagram of a first-in, first-out (FIFO) buffer of an adaptive multi-rate (AMR) decoder.
  • 13 to FIG. 16 are schematic diagrams of a mobile station processing five consecutive adaptive multi-rate (AMR) coded frames;
  • FIG. 13 is a schematic diagram of transmitting a no-data (NO-DATA) frame to a first-in-first-out (FIFO) buffer;
  • 14 is a schematic diagram of a first-in-first-out (FIFO) buffer transmission adaptive multi-rate (AMR) frame to variable rate decoder after receiving a no-data (NO_DATA) frame;
  • Figure 15 is a schematic diagram of the first in first out (FIFO) buffer after receiving two 10 millisecond transmission time intervals ( ⁇ ) of adaptive multi-rate (AMR) frames;
  • Figure 16 is a diagram of a first-in, first-out (FIFO) buffer that deletes a no-data (NO-DATA) frame after receiving two 10 ms transmission time interval (TTI) adaptive multi-rate (AMR) frames.
  • FIFO first-in, first-out
  • FIG. 17 is a block diagram of an embodiment of a network side generating and transmitting an adaptive multi-rate (AMR) coded frame.
  • AMR adaptive multi-rate
  • Figure 18 is a block diagram of an embodiment of a network side transmitting an adaptive multi-rate (AMR) coded frame in an intermittent manner and a network side transmitting delayed adaptive multi-rate (AMR) coded frame.
  • AMR adaptive multi-rate
  • AMR delayed adaptive multi-rate
  • FIG 19 is a basic functional block diagram of a Universal Mobile Telecommunications System (UMTS) handset. detailed description
  • UMTS Universal Mobile Telecommunications System
  • Embodiment 1 Mobile Station Generation and Transmission of Adaptive Multi-Rate (AMR) Coded Frames
  • an adaptive multi-rate (AMR) encoder converts voice from sound to an electrical signal, and after filtering, is sampled to convert from an analog voice signal to a digital voice signal 1.
  • the encoding module encodes the digital voice signal 1 in a manner of 20 milliseconds one frame.
  • the encoding module generates one voice frame coding sequence 2 every 20 milliseconds.
  • Each voice frame coding sequence includes: mode indication, several adaptive multi-rates (AMR) A voice coded frame or an adaptive multi-rate (AMR) silence frame.
  • the mode indication gives the case of the voice frame coding sequence and its adaptive multi-rate (AMR) coded frame, SP, whether it is a valid voice frame coding sequence, and the adaptive multi-rate (AMR) coded frame in the voice frame coding sequence. Number and encoding mode for each frame.
  • the transport format combination selection module in the wireless interface functional unit uses the voice frame coding sequence 2 of the adaptive multi-rate (AMR) voice coder and The packet service data 4 and the signaling data 5 are inputs, a transport format combination is selected, and the transport format of the transport voice and the voice data transport block 6 corresponding to the format are determined according to the transport format combination.
  • the voice data transmission block 6 is an adaptive multi-rate (AMR) coded frame, and the adaptive multi-rate (AMR) coded frame is included in the voice frame coding sequence 2.
  • AMR adaptive multi-rate
  • the transport blocks of the corresponding packet service data 4 and the signaling data 5 output by the transport format combination selection module are the packet data transport block 8 and the signaling data transport block 7, respectively.
  • the other modules of the wireless interface function unit map the voice data transmission block 6, the packet data transmission block 8, and the signaling data transmission block 7 onto the physical channel for transmission.
  • the wireless interface function unit has a voice frame mode control module for outputting the mode selection signal 9 to the adaptive multi-rate (AMR) encoder, the mode selection signal 9 is output to the encoding module, and the mode selection signal 9 includes a plurality of adaptive multi-rates.
  • the mode selection signal 9 as an encoding command indicates that the encoding module should include a plurality of adaptive multi-rate (AMR) speech encoded frames when outputting non-silent frames, and indicate the mode of these frames.
  • Table 1 is an example of the composition of a voice frame coding sequence 2 and its bit number allocation.
  • the digital voice signal 1 is simultaneously output to the voice encoding function module 100 and the voice encoding module 101 with voice activation detection.
  • the voice encoding function module 100 of the voice activation detection has substantially the same structure as the corresponding portion of the Overview of audio processing fimctions of the Universal Mobile Telecommunications System (UMTS) standard TS26.071 (or TS 26.171). The difference is that the coding mode indication signal 17 output to the speech coding sub-module of the speech coding function module 100 with voice activation detection is shown in FIG.
  • UMTS Universal Mobile Telecommunications System
  • the voice activation detection module outputs the voice A voice activity detection Detector flag 10, a voice coding module of the voice coding function module 100 with voice activation detection outputs a voice coding frame 12, and the voice coding module 101 outputs a voice coding frame 19, and voice coding frames 12 and 19
  • the number of bits per frame depends on the coding mode indication signals 17 and 18, respectively, and the coding mode control signals 17 and 18 are two outputs of one mode one way in which the coding mode control module decomposes the two modes of the mode selection signal 9.
  • the discontinuous transmission and operation module outputs an adaptive multi-rate (AMR) frame type signal 11 to a multi-channel speech coding multiplexing module and a speech coding module, and an adaptive multi-rate (AMR) frame type signal 11 indicates: whether the information bit 14 is valid Adaptive multi-rate encoded frame, mode of adaptive multi-rate encoded frame, mode of the adaptive multi-rate encoded frame is a mode of a non-silent speech frame or a mode of a silent frame, when the mode of the adaptive multi-rate encoded frame is When the frame is silenced, the information bit 14 is the silence detection frame 13 output by the comfort noise transmitting module.
  • AMR adaptive multi-rate
  • the speech encoding module 101 outputs an adaptive multi-rate speech encoded frame 19 and its adaptive multi-rate (AMR) frame type signal 16, which represents the mode of the adaptive multi-rate speech encoded frame 19.
  • the multiplexed speech coding multiplexing module combines the information bits 14 from the discontinuous transmission and operation module and the adaptive multi-rate speech coding frame 19 into a speech frame coding sequence 2, the method of combining is: when adaptive multi-rate (AMR) frames When the type signal 11 indicates that the information bit 14 is invalid, the mode of the voice frame coding sequence 2 is set to be an invalid voice frame coding sequence; when the adaptive multi-rate (AMR) frame type signal 11 indicates that the information bit 14 is a silent frame, the voice is set.
  • AMR adaptive multi-rate
  • the mode of the frame coding sequence 2 is indicated as a silence frame, and the information bit 14 is placed in the voice frame coding sequence of the voice frame coding sequence 2; when the adaptive multi-rate (AMR) frame type signal 11 indicates that the information bit 14 is a non-silent frame, The information bits 14 and 19 are placed together in the voice frame coding sequence of the i-tonal frame coding sequence 2, while the mode indication of the voice frame coding sequence 2 is set to be indicated by the adaptive multi-rate (AMR) frame type signals 11 and 16. mode.
  • AMR adaptive multi-rate
  • Table 2 shows the composition of information bits 14 and adaptive multi-rate speech encoded non-silent frames 19 generated by the encoding module when the number of adaptive multi-rate (AMR) modes in an encoding command is 2, packet service data 4 and signaling data 5
  • AMR adaptive multi-rate
  • Transport channel 1 2 3 1 2 3 5 4 Logical channel carrying user data
  • Table 3 shows the attributes and parameters of each transport channel in Table 2, in particular the configured transport format (TF).
  • the configured transport format supports adaptive multi-rate (AMR) voice coder modes of 12.2 kbps, 7.4 kbps and 4.75 kbps.
  • the voice frame mode control part of the wireless interface function unit sends out to the AMR voice coder.
  • the mode selection signal 9 may include any two of the above three modes, for example, (12.2 kbps, 4.75 bps).
  • Table 4 gives the meaning of all combinations of transport formats, that is, the user data that the transport format combination is used to transmit.
  • Table 4 Transport Channel TrChl TrCh2 TrCh3 TrCh4 TrCh5 crown,
  • TFI 0 0 0 0 0 0 does not send data
  • the radio interface functional unit is to process voice data of 12.2 kbps and 4.75 bps in the voice frame coding sequence 2, if There is no valid data in the output of the transport channel signaling data 5 and the packet data 6 to be scheduled, and the transport format combination (2, 1, 1 , 0, 0) is a valid combination, and the wireless interface functional unit can transmit the mode at 12.2 kbps.
  • Voice frame if there is valid data to be transmitted in the transmission channel signaling data 5, there is no data in the packet data, and the transmission format combination (2, 1, 1, 0, 1) is limited due to the limitation of the physical channel bandwidth. It is an invalid combination, and the transport format combination (4, 3, 0, 0, 1) is not effectively limited by the bandwidth limitation of the physical channel.
  • the wireless interface functional unit can simultaneously transmit the voice frame and signaling with the mode of 4.75 kbps. .
  • the radio interface functional unit is to process voice data of 12.2 kbps and 4.75 bps in the voice frame coding sequence 2, and the configuration of the physical channel changes if J3 ⁇ 4 , causing the bandwidth of the channel to decrease such that the transport format combination (2, 1, 1, 0, 0) becomes an invalid combination, and the transport format combination (3, 2, 0, 0, 0) and (4, 3, 0, 0, 0) The bandwidth limit without the physical channel is a valid combination.
  • the wireless interface function unit transmits a voice frame of 4-75 kbps mode when only voice data is available, and transmits the content to the AMR voice coder (7.4 kbps).
  • Mode selection signal 9 of 4.75 bps after changing to 7.4 kbps and 4.75 bps of voice data in the voice frame coding sequence 2, the wireless interface function only when voice data is available The unit can transmit a voice frame of mode 7.4 kbps.
  • FIG. 3 is a schematic diagram of processing of an encoding module having a number of adaptive multi-rate (AMR) modes of 3 included in the mode selection signal 9.
  • the digital voice signal 1 simultaneously transmits a voice encoding function module 100 with a voice activation detection, and a voice encoding module 101.
  • the speech encoding module 102 outputs, the voice encoding function module 100 with voice activation detection is in the Overview of audio processing functions of the Universal Mobile Telecommunications System (UMTS) standard TS26.071 (or TS 26.171).
  • UMTS Universal Mobile Telecommunications System
  • TS26.071 or TS 26.171
  • the structure of the corresponding part is basically the same, the difference is that the coding mode indication signal 17 outputted to the speech coding module is shown in FIG.
  • the detected speech encoding function module 100 has a speech encoding sub-module that outputs a speech encoding frame 12, and the speech encoding module 101 and the speech encoding module 102 output a separate LI that is a speech encoding frame 19 and 21, and each of the speech encoding frames 12, 19, and 21
  • the number of bits depends on the coding mode indication signals 17, 18 and 15, respectively.
  • a signal indicative of formula 17, 18 and 15 are the encoding mode selected in the mode control module 9 3 3 mode signal into an output channel pattern 1.
  • the discontinuous transmission and operation module outputs an adaptive multi-rate (AMR) frame type signal 11 to a multi-channel speech coding multiplexing module and a speech coding module, and an adaptive multi-rate (AMR) frame type signal 11 indicates: whether the information bit 14 is valid Adaptive multi-rate coded frame, mode of adaptive multi-rate coded frame, mode of the adaptive multi-rate coded frame is a mode of a voice frame or a mode of a silence frame; when the mode of the adaptive multi-rate coded frame is silent At the time of frame, the information bit 14 is the silence detection frame 13 output by the comfort noise transmitting module.
  • AMR adaptive multi-rate
  • the speech encoding module 101 and the speech encoding module 102 output adaptive multirate speech encodings 19 and 21 and respective adaptive multirate (AMR) frame type signals 16 and 20, respectively, which represent adaptive multirate speech encoding frames 19 and 21, respectively.
  • the multiplexed speech coding multiplexing module combines the information bits 14 from the discontinuous transmission and operation modules with the adaptive multi-rate speech coding frames 19 and 21 into a speech frame coding sequence 2, the combining method being: when adaptive multi-rate (AMR) When the frame type signal 11 indicates that the information bit 14 is invalid, the mode of the voice ⁇ code sequence 2 is set to be an invalid voice frame code sequence; when the adaptive multi-rate (AMR) frame type signal 11 indicates that the information bit 14 is a silence frame, The mode of the voice frame coding sequence 2 is indicated as a silence frame, and the information fc is placed in the voice frame coding sequence of the voice frame coding sequence 2; when the adaptive multi-rate (AMR) frame type signal 11 indicates that the information bit 14 is
  • Embodiment 2 - The mobile station transmits an adaptive multi-rate (AMR) coded frame in an intermittent manner
  • AMR adaptive multi-rate
  • FIG. 4 An embodiment of a method for generating and transmitting an adaptive multi-rate (AMR) voice frame with a buffered mobile station is shown in FIG. 4.
  • the difference between FIG. 4 and FIG. 1 is that the first-in first-out memory (FIFO) is combined with the transport format.
  • the ED first-in memory read-out control component replaces the transport format combination selection component of FIG.
  • the output voice frame coding sequence 2 of the encoding module is to the first in first out memory (FIFO), the first in first out memory (FIFO) buffers the voice frame code sequence, and outputs the storage status flag 25, and the storage status flag 25 indicates: whether there is any unread The voice frame coding sequence, and the number of these unread voice frame coding sequences.
  • Transport format combination selects first-in-first-out storage read control
  • the component outputs a read command 26, and the read command 26 causes the first in first out memory (FIFO) to output a live frame of the sequence 3 .
  • the transport format combination selects the first in first out memory read control component to check the number of voice fe3 ⁇ 4 code sequences stored in the first in first out memory (FIFO) given by the storage status flag 25. Whether the limit value is exceeded, when it is exceeded, the voice frame coding sequence with timeout storage is determined, and then the time-out voice frame coding sequence is taken out from the first-in first-out memory (FIFO) by the read command 26, and can be discarded by delay control.
  • the above limit value is determined to the extent.
  • Table 5 shows the configuration of an exemplary transport channel in the course of operation, giving the attributes and parameters of each transport channel, in particular the transport format (TF) D transmission corresponding to the transport format identifier (TFI) of each transport channel.
  • TTI Time interval
  • the voice frame coding sequence stored in the first-in-first-out memory (FIFO) of the radio interface function fetcher contains voices of 12.2 kbps and 4.75 bps modes.
  • Data if there is valid data to be transmitted in the transmission channel signaling data 5, there is no valid data in the dividend data, and the transmission format combination (2, 1, 1, 0, 1) is none due to the limitation of the physical channel bandwidth.
  • the wireless interface function unit can use the transmission interface format (0, 0, 0, 0, 1) after transmitting the signaling data in a 20 millisecond transmission day interval, the wireless interface function unit can use (4, 3 , 0, 0, 0)
  • the buffered voice coded frame with the mode of 4.75 kbps is transmitted in two consecutive 10 millisecond transmission time intervals. The previous one of the two is delayed due to preemption, and the latter is undelayed. In this way, the voice data temporarily suspended by the signaling data is delayed to be transmitted without affecting the subsequent transmission of the voice frame. If the above preemption is a silent start frame, the method of transmitting the silenced start frame can be sent (5, 0, 0, 0, 0) to delay transmission.
  • the time limit of the first-in first-out memory is set to 10 mobile stations with a length of 20 milliseconds.
  • the wireless interface function unit cannot send the wireless frame.
  • the switch completes the wireless interface function unit,
  • eight voice frame coding sequences are stored in the first in first out memory (FIFO) during the switching. These voice frame coding sequences are the same as the previous example.
  • the mode selection signal 9 contains the mode (12.2 kbps, 4.75 bps).
  • the wireless interface function unit can transmit (4, 3, 0, 0, 1) an adaptive voice coded frame of mode 4.75 kbps and transmit 16 (8+4+2+1+1) from the current frame.
  • the delayed voice frame coding sequence stored in the FIFO for more than 20 milliseconds can be recovered after 20 milliseconds of the voice coded frame of mode 12.2 kbps.
  • the combination of the voice frame mode control portion and the transport format combination in FIG. 4 selects the transport format combination in the FIFO read control portion to be implemented on the central processing unit, and selects the transport format combination to select the FIFO memory read control portion.
  • the FIFO read control is placed in the channel encoder to implement an embodiment of the interface buffered in the mobile station shown in FIG.
  • the first in first out (FIFO) buffer is between the adaptive multi-rate (AMR) encoder and the channel encoder, and the adaptive multi-rate (AMR) encoder outputs one to the first in first out (FIFO) buffer every 20 milliseconds.
  • the voice frame coding sequence, the first in first out (FIFO) buffer in the figure is used to store the voice frame coding sequence, and the central processor reads out the type of the voice frame coding sequence to be processed through the interface 29 shown in the figure, the central processor
  • An encoding command comprising a plurality of adaptive multi-rate (AMR) modes can be transmitted to the adaptive multi-rate (AMR) encoder via the interface 28 shown in the figure.
  • Table 6 gives an example of a combination of transport channel formats based on the sequence of voice frame codes written in the first-in, first-out (FIFO) buffer during the period from 0 to 80 milliseconds in Table 6.
  • TFC Transport Channel Format Combination
  • FIFO milliseconds buffer code sequence and its type.
  • AMR Multi-rate
  • TTI frame code sequence
  • mode 1+ mode 7 (81,103,60), 20 ms mode 7
  • mode 1+ mode 2 (55, 63), 10 ms mode 2
  • FIG. 8 shows the 0 millisecond time in the first in first out (FIFO) buffer stored in the voice frame coding sequence 305, as shown in Table 6, this
  • the sequence of the speech frame to be processed read by the central processing unit through the interface 29 shown in the figure is a valid voice frame coding sequence 305 of the type 1 + mode 7, the combination of the transmission channel format on the central processing unit.
  • the result of the (TFC) selection is a combination of a transmission time interval (TTI) of 20 milliseconds, which contains three formats for the voice transmission channel, one format being 1 X 89 bits and the other being 1 X 103 bits. There is also a 1 X 60 bit.
  • TTI transmission time interval
  • the adaptive multi-rate (AMR) frame of mode 7 in the voice frame coding sequence 305 is to be placed on the transmission channel, 501 in Figure 7 is the adaptive multi-rate (AMR) frame of mode 7.
  • the specific operation is such that the central processor issues an instruction to the channel coder via the interface 27 to encode according to a combination of transmission formats including 3 X 89 bits, 1 X 103 bits and 1 X 60 bits. It will be operative to read the adaptive multi-rate (AMR) encoded frame 501 of mode 7 in the first in first out (FIFO) buffered speech frame encoding sequence 305, which shifts the speech frame encoding sequence 305 out of the first in first out (FIFO) Cache.
  • FIFO first in first out
  • Table 20 shows that at 20 milliseconds, the central processor reads out the voice frame coding sequence to be processed by the interface 29 shown in the figure as a valid voice frame coding sequence 304 of the type 1 + mode 7, transmission channel format.
  • the result of the combination (TFC) selection is that there is no transport format combination that can transmit voice. There are many reasons for this (physical channel change, high priority logical channel, and handover), so this is not the first in first out (FIFO).
  • the cache reads the voice frame encoding sequence 304.
  • Figure 8 shows a voice frame encoding sequence 304 stored in a first in first out (FIFO) buffer at 35 milliseconds, while an adaptive multi-rate (AMR) encoder outputs a voice frame encoding sequence 303, a voice frame to a first in first out (FIFO) buffer.
  • the type of the coding sequence 303 is mode 1 + mode 2, which is changed compared to the previous speech frame coding sequence 304, i.e., the mode change in the coding command of the central processor is reflected in the speech frame coding sequence 303.
  • Figure 9 shows the voice frame encoding sequences 304 and 303 stored in the first in first out (FIFO) buffer at 40 milliseconds, as shown in Table 6, at this time because of the number of voice frame encoding sequences stored in the first in first out (FIFO) buffer.
  • a transmission format combination exceeding '1, preferably 10 milliseconds, of transmission time interval (TTI), at which point the result of the Transport Channel Format Combination (TFC) selection is a combination of a transmission time interval ( ⁇ ) of 10 milliseconds, which is included for
  • TTI transmission time interval
  • TFC Transport Channel Format Combination
  • 401 in FIG. 10 is the adaptive multi-rate (AMR) frame of the mode 1, and the specific operation is such that the central processor sends the channel encoder through the interface 27 according to the two of the X X49 bits and the 1 X 54 bits.
  • Voice transmission format transmission The format combines the encoded instructions, which will cause it to issue an operation of reading the adaptive multi-rate (AMR) encoded frame 401 of mode 1 in the first in first out (FIFO) buffered speech frame encoding sequence 304, which will sequence the speech frame encoding 304, remove the first in first out (FIFO) cache.
  • AMR adaptive multi-rate
  • the voice frame coding sequence 302 written at 55 milliseconds as shown in Table 6 is not read until 70 milliseconds, the write and read time intervals exceed 10 milliseconds, since the Transport Channel Format Combination (TFC) selection can be used to shorten the The write and read time intervals are described, so the result of the Transmission Channel Format Combination (TFC) selection at 70 milliseconds is the 10 ms transmission time interval (the combination of the transport format of the TTD, ie the table shown in the table "(55, 63), 10 milliseconds", this will result in a significant reduction in the time interval between the read operation at 80 milliseconds and the write operation at 75 milliseconds compared to the last pair of operations.
  • TFC Transmission Channel Format Combination
  • Embodiment 3 Mobile Station Delay Adaptive Multi-Rate (AMR) Decoding
  • the mobile station's receive source control rate processor (Rx SCR handler) is associated with the radio interface functional unit through a first in first out memory (FIFO), 32, 33 and 34 are first in first out memory (FIFO) directions, respectively.
  • the receiving bit of the mobile station controls the information bits, mode indication and reception type of each received frame output by the Rx SCR handler, 37, 38 and 39 are respectively the wireless interface function unit to the first in first out memory (FIFO)
  • the information bits, mode indication and reception type of each received frame are output.
  • the mode indication gives the adaptive multi-rate of the received frame (AMR mode, the reception type is shown in Table 7.
  • the first in first out memory (FIFO) will be the wireless interface function.
  • Each adaptive multi-rate (AMR) receive frame sent by the unit is sequentially cached, and the above information bits, mode indication and reception type are saved together during buffering.
  • the cache status flag 30 indicates: whether there is an unread adaptive Multi-rate (AMR) received frames, and the number of these unread adaptive multi-rate (AMR) received frames.
  • AMR adaptive Multi-rate
  • the R SCR handler does not immediately read and decode the received unread received frame, but is not read in the first in first out memory (FIFO).
  • FIFO first in first out memory
  • the mobile station's receive source control rate processor SCRhandler begins to read an adaptive from every 20 milliseconds.
  • the multi-rate encoded frame is read by issuing a read command 31 to the first in first out memory (FIFO).
  • the first in first out memory (FIFO) output receives the lost status flag 35, which gives the mobile station in speech (SPEECH) mode.
  • the number of actively added receive lost frames (of type NO_DATA_SPEECH).
  • the value given by Receive Loss Status Flag 35 indicates that the value added in voice (SPEECH) mode stored in the First In First Out Memory (FIFO) is actively added.
  • the wireless interface function unit no longer adds the received loss to the first-in-first-out memory (FIFO) output voice (SPEECH) mode. Frame loss.
  • the wireless interface function unit If the wireless interface function unit outputs a receive frame to the first in first out memory (FIFO) at a rate of more than 1 frame every 20 milliseconds, the wireless interface function unit sends a delete command 36 to the first in first out memory (FIFO) to be first in, first out. Receive lost frames deleted in voice (SPEECH) mode stored in the memory (FIFO).
  • SPEECH Receive lost frames deleted in voice
  • Table 8 shows a series of operations on the received frame delayed by the mobile station.
  • the first-in-first-out memory (FIFO) buffer has a lower limit of 4, and its speech (SPEECH) mode. Add and receive The specified limit value for the number of lost frames is 2.
  • the processing of the received frames in the first 20 milliseconds to the eighth 20 millisecond time period of each unit in FIG. 11 is listed in the table.
  • SPEECH GOOD Empty SPEECH mode
  • wireless interface The empty function unit writes the first frame to the first-in first-out memory, and the Rx SCR handler does not read the frame.
  • the wireless interface function unit writes the empty second frame to the FIFO memory, and the Rx SCR handler still does not read the frame.
  • the wireless interface function unit writes the third frame of the third frame to the FIFO storage. Empty, Rx SCR handler still does not read frames
  • Fourth frame second frame first frame mobile station is still in SPEECH mode
  • the wireless interface function unit writes the fourth frame of H fourth frame, the number of the first-in first-out memory is up to 4, and the Rx empty SCR handler starts reading every 20 milliseconds.
  • the wireless interface function unit writes the sixth frame, the sixth frame, the fifth and sixth frames, deletes
  • SPEECH GOOD Empty Fourth Empty Frame in FIFO Memory Receive Lost Frame Added in SPEECH Mode
  • Seventh frame, fifth frame, third frame, mobile station is still in SPEECH mode
  • the wireless interface function unit writes the seventh frame of H seventh frame
  • the eighth frame, the sixth frame, the fifth frame, the mobile station is still in the SPEECH mode.
  • NO-DATA-SPEEC The eighth frame type, the first-in first-out memory has been empty. There are two received loss frames added in the SPEECH mode, and the ninth frame of the ninth frame added in the SPEECH mode is no longer written.
  • the eleventh frame tenth frame wireless interface function unit puts the tenth
  • SPEECH GOOD
  • the eighth and eleventh frames are written to the eleventh frame in the empty SPEECH mode.
  • SPEECH GOOD receives the lost frame in front of the frame, deletes the seventh frame in the FIFO memory, adds the received loss frame in SPEECH mode, and the R SCR handler reads the tenth frame.
  • SPEECH-GOOD writes the thirteenth frame two and thirteenth frames, deletes the eighth SPEECH-GOOD frame in the thirteenth frame empty first-in first-out memory, the received lost frame added in the SPEECH mode, and the Rx SCR handler reads the first ⁇ —frame
  • SPEECH—GOOD is written in the fourteenth frame and the fifteenth frame. Because the fifteenth frame is not preceded by the SPEECH GOOD empty received loss frame in SPEECH mode, there is no delete operation, and the Rx SCR handler reads the first. Twelve Frames Figure 12 shows a block diagram of the above-described delay decoding method.
  • the radio interface function in Figure 11 is implemented in the channel decoder and the central processor.
  • the mobile station receives the source control rate processor and puts it on.
  • the variable rate decoder and the central processor are implemented.
  • the first in first out (FIFO) buffer is between the variable rate decoder and the channel decoder, the channel decoder performs decoding operations at 10 millisecond frame timing, and the central processor receives the adaptive multi-rate generated by the channel decoder via interface 54 ( AMR) frame message, the central processor can send an instruction to the channel decoder to output an adaptive multi-rate (AMR) frame to the first in first out (FIFO) buffer via interface 54; the central processor can read the first in first out through interface 53 (FIFO) The length of the adaptive multi-rate (AMR) encoded frame queue stored in the buffer and the type of these encoded frames. Every 20 milliseconds, the central processor issues a delete to the first in first out (FIFO) buffer through interface 53. The central processor may issue an instruction to read the adaptive multi-rate (AMR) frame to the variable rate decoder via interface 58.
  • AMR adaptive multi-rate
  • the radio frame of the (AMR) frame transmits an unadaptive multi-rate (AMR) frame indication to the central processor via interface 54, and the central processor sends an output no data (N0JMTA) frame 52 to the channel decoder via interface 54 to first in first out.
  • AMR unadaptive multi-rate
  • N0JMTA output no data
  • FIFO cached instructions
  • AMR adaptive multi-rate
  • AMR adaptive multi-rate
  • the first in first out (FIFO) buffer reads the instructions to the adaptive multi-rate (AMR) decoder, as shown in Figure 14, after receiving the instruction, the adaptive multi-rate (AMR) decoder is from the first in first out (FIFO)
  • An adaptive multi-rate (AMR) encoded frame 56 read out of the buffer.
  • the next two 10 millisecond frame timing channel decoders decode two adaptive multi-rate (AMR) frames 51 and 50 and pass through interface 54 to the central processor after each adaptive multi-rate (AMR) frame decoding is completed.
  • the transmit generates an adaptive multi-rate (AMR) frame message, and each time the message is received, the central processor sends an adaptive multi-rate (AMR) frame to the first in first out (FIFO) buffer via the interface 54 to the channel decoder.
  • Figure 15 shows the situation of frames stored in a first in first out (FIFO) buffer after receiving these two adaptive multi-rate (AMR) frames.
  • the central processing unit acquires a first in first out (FIFO) buffer through the interface 53 to store information of the no data (NO-DATA) frame 52, issues an instruction to delete the no data (NO-DATA) frame, and executes the first in first out after the instruction.
  • FIFO first in first out
  • the situation of the frames stored in the buffer is as shown in FIG. 16.
  • Embodiment 4 Network Side Generation and Transmission of Adaptive Multi-Rate (AMR) Coded Frames
  • a transcoder (TC) speech encoded signal 41 is transcoded to produce a speech frame encoding sequence 42, each speech frame encoding sequence comprising a number of adaptive multi-rate (AMR) speech encodings.
  • a frame or an adaptive multi-rate (AMR) silence frame or a no-data (NO-DATA) type frame, each frame in the voice frame coding sequence 42 includes not only information bits of an adaptive multi-rate (AMR) core frame.
  • encoder cyclic redundancy check (CRC:) frame type, quality indication, and mode indication.
  • the voice frame coding sequence 42 generated by the pattern converter (TC) is processed by the Iu interface function unit to form an Iu user plane frame 43 and input to the radio network controller (RNC), and then restored to a voice frame by the Iu interface function unit.
  • the code sequence 42 is then output to the transport format combination selection unit, and the frame type of each frame in the voice frame coding sequence 42 indicates whether the frame is a voice frame or a silence frame or no data.
  • the mode indication of each frame in the voice frame coding sequence 42 gives the coding mode of the frame.
  • the transport format combination selection module in the radio network controller (RNC) takes as input a voice frame coding sequence 42, packet service data 44, and signaling data 45, and the transport format combination selection module selects a transport format combination, and combines according to the transport format. Determine the transport format of the transmitted voice and the corresponding output transport block.
  • the frame in the voice frame coding sequence 42 is a voice frame or a silence frame instead of a no-data (NO-DATA) type frame
  • the corresponding transport block is an adaptive multi-rate.
  • AMR adaptive multi-rate
  • AMR adaptive multi-rate
  • AMR adaptive multi-rate
  • the corresponding transport blocks of the packet service data 44 and the signaling data 45 are a packet data transport block 48 and a signaling data transport block 47, respectively.
  • the other units of the Radio Network Controller (RNC) map the voice data transport block 46, the packet data transport block 48, and the signaling data transport block 47 onto the physical channel for transmission.
  • Radio Network Controller (RNC) Media Gateway to the Core Network
  • the rate control unit of the media gateway (MGW) generates a mode selection signal 40 indicating a plurality of adaptive multi-rate (AMR) voice coding modes according to the RFCI-RAB sub-Flow Combination Indicator (RFCI-RAB sub-Flow Combination Indicator).
  • the mode select signal 40 is output to a pattern converter (TC) specifying a pattern of a plurality of adaptive multi-rate (AMR) coded frames that should be included in the sequence of speech encoded frames 42 output by the pattern converter (TC).
  • Radio access bearer substream combination indication (RFCI)
  • RFCI Radio Access Bearer Substream Combination
  • Table 9 is an example of a Radio Access Bearer Substream Combination (RFC) used by a Radio Network Controller (RNC) to transmit a voice frame coding sequence to a core network and its Radio Access Bearer Subflow Group Indicator (RFCI) values, which are also The part of the core network-radio access network interface user plane (Iu UP) initialization process that needs to be sent during Iu UP initialization.
  • RRC Radio Access Bearer Substream Combination
  • RNC Radio Network Controller
  • RFCI Radio Access Bearer Subflow Group Indicator
  • the numbering identifier (RFCI n indicator) of the radio access bearer substream combination indication using the core network-line access network interface (Iu) user plane rate control frame 49 is given to the core network.
  • the media gateway (MGW) issues an indication corresponding to the RFC, and the rate control portion determines a plurality of adaptive multi-rate (AMR) coding modes according to the indication, and outputs a mode selection including the plurality of adaptive multi-rate (AMR) coding modes.
  • AMR adaptive multi-rate
  • the Access Bearer Substream Combination Indicator carries the frames in the voice frame encoding sequence 42.
  • Embodiment 5 The network side transmits an adaptive multi-rate (AMR) coded frame in an intermittent manner
  • AMR adaptive multi-rate
  • the scheme of FIG. 18 is a modification of FIG. 17, which differs from FIG. 17 in that a FIFO is added and a FIFO read control component is selected in combination with a transport format instead of FIG.
  • the transport format combination selects the components.
  • the voice frame coding sequence 42 outputted by the pattern converter is processed as shown in FIG. 17, and is processed by the Iu interface function unit to form an Iu user plane frame 43 and input to the radio network controller (RNC), and then processed by the Iu interface function unit.
  • RNC radio network controller
  • the transport format combination selects the first in first out memory read control unit output read command 422, and the read command 422 makes the advanced The first-out memory (FIFO) outputs a voice frame coding sequence 420.
  • the first in first out memory read control component can read and discard the voice frame encoding sequence from the first in first out memory (FIFO) using the read command 422.
  • Table 11 shows the configuration of an exemplary transport channel in the working process of the embodiment, giving the attributes and parameters of each transport channel, in particular the transport format (TF) corresponding to the transport format identifier (TFI) of each transport channel. And transmission time interval ( ⁇ ).
  • Table 9 is used as an example of a Radio Access Bearer Substream Combination (RFC) and its Radio Access Bearer Substream Combination Indicator (RFCI) value, when the Core Network-Radio Access Network Interface (Iu) User Plane Rate Control Frame 49
  • the sequence identifier (RFCI n indicator) indicated by the radio access bearer substream combination is (0, 1, 2, 4, 6), so that the mode selection signal 40 output by the rate control unit includes a mode (12.2 kbps, 4.75 bps).
  • the core network transmits the voice frame coding sequence 42 using the RFCI with a value of 6 in the core network-radio access network interface (Iu) user plane frame, and the radio network controller (RNC) is stored in the first in first out memory (FIFO).
  • the voice frame coding sequence contains voice data in the 12.2 kbps and 4.75 bps modes. If there is valid data to be transmitted in the transmission channel signaling data 5, there is no valid data in the packet data, and due to the limitation of the physical channel bandwidth.
  • the transport format combination (2, 1, 1, 0, 1) is an invalid combination, so that the radio interface function unit sends the message with a transport format combination (0, 0, 0, 0, 1) within a 20 ms transmission time interval.
  • the wireless interface function unit can use (4, 3, 0, 0, 0) to transmit a voice frame of 4.75 kbps in consecutive 2 10 millisecond transmission time intervals, so that the voice is suspended due to the preemption of the signaling data.
  • the data is delayed in transmission without affecting the subsequent transmission of voice frames.
  • the radio network controller cannot transmit the radio frame when the handover occurs, and the radio network controller receives a total of 8 voice frame coding sequences from the core network during the handover, the wireless network The controller determines 5 (4+1) as the timeout parameter, and discards the timeout sequence when the stored status flag 421 gives the number of stored voice frame coding sequences greater than 5, and these voice frame coding sequences are the same as the previous example, ( 12.2 Kbps, 4.75bps) mode, after the switch is completed, the wireless interface function unit can use (4, 3, 0, 0, 0) to transmit the pattern in the sequence of voice coded frames during the switching interval of 5 consecutive 10 milliseconds.
  • Embodiment 6 Network side transmission delayed adaptive multi-rate (AMR) coded frame
  • the embodiment 5 indicated in FIG. 18 can also be used for processing the core network to transmit a sequence of voice coded frames with delay flags to the radio network controller (RC), and the delay flag can be placed on the core network-radio access network interface (Iu) user.
  • the sequence of voice coded frames with delay flags is caused by the delayed adaptive multi-rate voice coded frame transmitted by the mobile station at the transmitting end.
  • the mobile station at the transmitting end is transmitting due to handover.
  • a plurality of voice frame coding sequences are accumulated in the buffer area, so that the adaptive multi-rate voice coded frames in the delayed voice frame coding sequences are transmitted after the handover is completed.
  • a sequence of voice coded frames with a delay flag always arrives at the Radio Network Controller (RNC) with other coded frame sequences within 20 milliseconds, so the Radio Network Controller (RC) does not send a single delay flag within 20 milliseconds.
  • the AMR encoded frame schedules the speech encoded frames with delay flags on the transport channel in a manner that multiple frames are transmitted at a transmission time of 20 milliseconds each.
  • Table 12 shows the configuration of an exemplary transport channel in the working process of the embodiment, giving the attributes and parameters of each transport channel, in particular the transport format (TF) corresponding to the transport format identifier (TFI) of each transport channel. And transmission time interval ( ⁇ ).
  • Table 13 is an example of scheduling a voice coded frame to a transmission channel made by an embodiment during operation.

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Abstract

The present invention discloses a method for adaptive multi-rate (AMR) coding and transporting speech in universal mobile transmission system (UMTS) and related mobile station, which allow the seamless AMR adjustment. Regardless at the mobile station or the network side, based on the multi-mode AMR coded frame generated from that speech is coded by multiple modes in coding order, and when deploying a speech frame to transporting channel, an AMR coded frame of some mode is selected from multi-mode AMR coded frame and is transmitted; then the network side can implement the said coding order through the rule of the core network-radio access network interface controlling multi-rate coding mode proposed by the present invention. According to the method of the present invention, buffering the AMR coded frame during the transmission, that is, in the mobile station setting first-in first-out (FIFO) buffer connected with the output of the adaptive speech coder, can allow not to discard the delayed AMR coded frames, which have not been transmitted by radio channel temporarily, and these delayed frames would be transmitted more rapidly after the radio channel is recovered.

Description

话音的自适应多速率编码和传输的方法和设备 技术领域  Method and device for adaptive multi-rate coding and transmission of speech
本发明涉及一种在通用移动通信*** (UMTS ) 中进行自适应和多速率的编码以及 传输所述编码产生的数据的方法和相关的移动台,具体涉及到改进自适应多速率(AMR) 编码器编码的方法和相关的移动台设备、 对输出到自适应多速率(AMR)解码器的数据 进行处理的方法和相关的移动台设备、无线接入过程中接入和控制自适应多速率(AMR) 编码帧的方法以及在中核心网 (CN) 和无线接入网 (RAN) 接口上传送自适应多速率 (AMR) 编码比特的方法。 技术背景  The present invention relates to a method for adaptive and multi-rate coding in a Universal Mobile Telecommunications System (UMTS) and for transmitting data generated by said coding, and related mobile stations, and in particular to improved adaptive multi-rate (AMR) coding Method of encoding and associated mobile station equipment, method of processing data output to an adaptive multi-rate (AMR) decoder, and associated mobile station equipment, access and control adaptive multi-rate during radio access ( AMR) A method of encoding frames and a method of transmitting adaptive multi-rate (AMR) coded bits on a medium core network (CN) and a radio access network (RAN) interface. technical background
移动台的结构如图 19所示, 其中的变速率话音编码器完成自适应多速率 (AMR) 编码的功能, 运行通信协议栈软件的中央处理器设置变速率话音编码器的模式, 并且负 责将传输格式组合(TFC) 选择的结果 (传输格式组合) 发送到信道编码器, 信道编码 器根据传输格式组合对信令数据、 来自变速率话音编码器的自适应多速率 (AMR) 编码 帧话音数据以及分组数据进行编码和复用处理, 信道编码器从变速率话音编码器放置其 生成的自适应多速率(AMR)编码帧的存储单元读取自适应多速率(AMR)编码帧; 信 道解码器完成信道解码生成自适应多速率(AMR) 编码帧, 变速率话音解码器从信道解 码器的放置自适应多速率(AMR)编码帧的存储单元读取自适应多速率(AMR)编码帧 进行译码。  The structure of the mobile station is as shown in FIG. 19, wherein the variable rate speech coder performs the function of adaptive multi-rate (AMR) encoding, and the central processor running the communication protocol stack software sets the mode of the variable rate speech coder, and is responsible for The result of the transport format combination (TFC) selection (transport format combination) is sent to the channel coder, and the channel coder combines the paired signaling data, adaptive multi-rate (AMR) encoded frame speech data from the variable rate speech coder according to the transmission format. And the packet data is encoded and multiplexed, and the channel coder reads the adaptive multi-rate (AMR) encoded frame from the storage unit of the adaptive multi-rate (AMR) encoded frame generated by the variable rate speech coder; the channel decoder Complete channel decoding to generate an adaptive multi-rate (AMR) coded frame, and the variable rate voice decoder reads the adaptive multi-rate (AMR) coded frame from the memory unit of the channel adaptive decoder's placed adaptive multi-rate (AMR) coded frame for translation code.
第三代移动通信伙伴计划 (3GPP) 的技术规范 (TS ) 对自适应多速率 (AMR) 编 码器作出如下的规定:  The Third Generation Mobile Communications Partnership (3GPP) Technical Specification (TS) specifies the Adaptive Multi-Rate (AMR) encoder as follows:
— 3GPP TS26.071中规定, 自适应多速率(AMR)编码器将一个精度为 13比特的脉 冲编码调制(PCM)格式的话音信号作为输入, 或者将其他格式的话音信号转换为 13比 特精度的脉冲编码调制(PCM)格式的话音信号作为输入。编码器的采样率为 8000样本 /秒, 编码器以 160个 13比特精度的脉冲编码调制(PCM)格式的样本组成的一个块为 单位进行编码, 产生编码模式为 12.2、 10.2、 7.95、 7.40、 6.70、 5.90、 5.15、 4.75 kbps 之一的话音编码帧或静默编码帧。  – 3GPP TS 26.071 specifies that an adaptive multi-rate (AMR) encoder takes a 13-bit pulse code modulation (PCM) format voice signal as input, or converts other format voice signals to 13-bit precision. A voice signal in a pulse code modulation (PCM) format is used as an input. The encoder has a sampling rate of 8000 samples/second. The encoder encodes a block of 160 samples of 13-bit precision pulse code modulation (PCM) format, resulting in encoding modes of 12.2, 10.2, 7.95, 7.40. A voice coded frame or a silence coded frame of one of 6.70, 5.90, 5.15, 4.75 kbps.
3GPP TS26.171中规定, 自适应多速率(AMR)编码器将一个精度为 14比特的脉冲 编码调制(PCM)格式的话音信号作为输入, 或者将其他格式的话音信号转换为 14比特 精度的脉冲编码调制(PCM)格式的话音信号作为输入。 编码器的采样率为 16000样本 /秒, 编码器以 320个 14比特的脉冲编码调制(PCM)格式的样本组成的一个块为单位 进行编码,产生编码模式为 23.85、 23.05、 19.85、 18.25、 15.85、 14.25、 12.65、 8.85、 6.6kbps 之一的话音编码帧或静默编码帧。 — 3GPP TS26.071和 TS26.171中规定, 自适应多速率(AMR)编码器可以按照一个 改变编码模式的命令每 20毫秒改变一次话音编码帧的比特速率。 3GPP TS 26.171 specifies that an adaptive multi-rate (AMR) encoder takes a 14-bit pulse code modulation (PCM) format voice signal as input, or converts other format voice signals into 14-bit precision pulses. A voice signal in a coded modulation (PCM) format is used as an input. The encoder has a sampling rate of 16,000 samples/second. The encoder encodes a block of 320 14-bit pulse code modulation (PCM) format samples, producing encoding modes of 23.85, 23.05, 19.85, 18.25, 15.85. , a voice coded frame of 14.25, 12.65, 8.85, 6.6 kbps or a silence coded frame. – 3GPP TS 26.071 and TS 26.171 specify that an adaptive multi-rate (AMR) encoder can change the bit rate of a voice coded frame every 20 milliseconds according to a command to change the coding mode.
— 3GPP TS26.101和 TS26.201中规定, 在自适应多速率(AMR)编码器输出的每一 帧的内容中, 应指明该帧的编码模式和自适应多速率 (AMR)编码器的指示编码模式和 请求编码模式, 并且帧的编码模式、 指示编码模式和请求编码模式都是唯一的。  – 3GPP TS 26.101 and TS 26.201 specify that in the content of each frame of the adaptive multi-rate (AMR) encoder output, the coding mode of the frame and the indication of the adaptive multi-rate (AMR) encoder should be indicated. The coding mode and the request coding mode, and the coding mode of the frame, the indication coding mode, and the request coding mode are unique.
通用移动通信*** (UMTS) 中的技术规范 3GPP TS25.415中附录 A中对话音呼叫 过程中的自适应多速率 (AMR) 编码帧在网络中的传输作出了提示性说明, 从中可以得 到以下结论:  Technical Specifications in the Universal Mobile Telecommunications System (UMTS) The adaptive multi-rate (AMR) coded frames in the voice call process in Appendix A of 3GPP TS 25.415 are presented in the network for illustrative purposes, from which the following conclusions can be drawn. :
一核心网 (CN) 以 RANAP协议同无线接入网 (RAN)进行信令消息的交互, 将无 线接入承载 (RAB)的属性传送给无线接入网(RAN),该属性包含了自适应多速率 (AMR) 20毫秒话音帧的各个模式的比特大小和每个模式中各个类的比特大小。  A core network (CN) exchanges signaling messages with the Radio Access Network (RAN) using the RANAP protocol, and transmits the attributes of the Radio Access Bearer (RAB) to the Radio Access Network (RAN), which includes adaptation. Multi-rate (AMR) The bit size of each mode of the 20 millisecond voice frame and the bit size of each class in each mode.
一无线网络控制器 (RNC)根据无线接入承载 (RAB) 的属性安排无线接入承载子 流组合指示(RFCI), RFCI同自适应多速率 (AMR) 20毫秒话音帧的模式对应。  A Radio Network Controller (RNC) schedules a Radio Access Bearer Substream Combination Indicator (RFCI) according to the Radio Access Bearer (RAB) attributes, which correspond to the mode of an Adaptive Multi Rate (AMR) 20 msec voice frame.
一无线网络控制器(RNC)根据 RFCI来配置协同工作的若干专用信道(DCH), DCH 中的一个传输格式(TF)同一个 RFCI相对应, 因此一个 RFCI对应一个或多个传输格式 组合标识 (TFCI), 所述 TFCI也就和自适应多速率 (AMR) 20毫秒话音帧的模式相对 应。  A radio network controller (RNC) configures a number of dedicated channels (DCHs) that cooperate to work according to the RFCI, and one transport format (TF) in the DCH corresponds to one RFCI, so one RFCI corresponds to one or more transport format combination identifiers ( TFCI), the TFCI also corresponds to the mode of an adaptive multi-rate (AMR) 20 millisecond voice frame.
一无线网络控制器 (R C) 发送核心网 -无线接入网接口初始化帧 (Iu initialisation frame ),通知核心网它所能向码型变换器提供的若干 RFCI及其关联无线接入承载 ( RAB ) 的子流大小。  A radio network controller (RC) transmits a core network-radio access network interface initialization frame (Iu initialisation frame) to inform the core network of several RFCIs and associated radio access bearers (RABs) that it can provide to the pattern converter. The size of the substream.
一移动台 (UE)配置协同工作的若干 DCH,在配置过程中需要无线网络控制器 (RNC) 和移动台 (UE)之间的 RRC信令消息的交互, 移动台 (UE)可根据传输格式(TF)告 知其语音编解码器收到和将要发送的话音的无线接入承载(RAB)子流(sub-flow)结构, 该无线接入承载(RAB) 子流 (sub-flow)结构就是自适应多速率 (AMR) 20毫秒话音 帧的各个类比特。  A mobile station (UE) configures several DCHs that work cooperatively, and requires interaction of RRC signaling messages between a radio network controller (RNC) and a mobile station (UE) during configuration, and the mobile station (UE) can be based on a transport format. (TF) informs the voice codec of the radio access bearer (RAB) sub-flow structure of the voice received and to be transmitted, the radio access bearer (RAB) sub-flow structure is Adaptive Multi-Rate (AMR) Class of bits for a 20 millisecond voice frame.
一将从核心网 -无线接入网 (Iu)接口初始化帧 (Iu initialisation frame) 中得到的若 干 RFCI及其关联无线接入承载(RAB) 的子流大小的信息送给码型变换器的编解码器 供配置使用。  A piece of RFCI and its associated radio access bearer (RAB) substream size information obtained from the core network-radio access network (Iu) interface initialization frame (Iu initialisation frame) is sent to the code converter The decoder is for configuration use.
一在核心网 -无线接入网接口发送含话音帧的业务数据单元 (SDU) 时, 将核心网- 无线接入网用户面(Iu UP)帧中 RPCI设置为同该自适应多速率(AMR) 20毫秒话音帧 的模式相对应的值;  When the service data unit (SDU) containing the voice frame is sent on the core network-radio access network interface, the RPCI in the core network-radio access network user plane (Iu UP) frame is set to be the same as the adaptive multi-rate (AMR). a value corresponding to the mode of the 20 millisecond voice frame;
在 RNC中, 用该 RFCI将核心网 -无线接入网用户面 (Iu UP)帧分解到协同工作的 若干 DCH上, 这些 DCH具有各自不同的比特保护级别;  In the RNC, the core network-radio access network user plane (Iu UP) frame is decomposed into a plurality of DCHs working in cooperation, and the DCHs have different bit protection levels;
无线帧的 TFCI由媒体接入层 MAC选择; 移动台 (UE) 从协同工作的若干 DCH上接收含有话音帧的 SDU, 将它们恢复成自 适应多速率(AMR) 20毫秒话音帧, 可以用 TFCI向话音编解码器指示话音帧的结构。 The TFCI of the radio frame is selected by the medium access layer MAC; The mobile station (UE) receives the SDUs containing the voice frames from a number of DCHs that work together and restores them to an adaptive multi-rate (AMR) 20 millisecond voice frame, which can be used to indicate the structure of the voice frame to the voice codec.
由上述规定可见,每个 20毫秒的话音输入被转换成某一个特定自适应多速率 (AMR) 模式的编码帧。  As can be seen from the above, each 20 millisecond voice input is converted into a coded frame of a particular adaptive multi-rate (AMR) mode.
通用移动通信*** (UMTS) 中的技术规范 3GPP TS25.944中对上述映射自适应多 速率 (AMR) 20毫秒话音帧的 DCH传输信道格式的配置给出了示例, 并在示例中对从 传输信道数据输入到无线帧分段作了描述, 具体描述在 3GPP TS25.944的 4.1.1.3.1.2和 4.2.1.3.1.2中可见。  An example of the configuration of the DCH transport channel format of the above-described map adaptive multi-rate (AMR) 20 msec voice frame is given in the technical specification 3GPP TS 25.944 in the Universal Mobile Telecommunications System (UMTS), and in the example the slave transport channel The data input to the radio frame segmentation is described in detail in the description of 4.1.1.3.1.2 and 4.2.1.3.1.2 of 3GPP TS 25.944.
通用移动通信*** (UMTS) ***中, 传输信道中的发送数据被送到复合信道, 经 过 CRC粘贴、 传输块级联 /码块分割、 信道编码等操作步骤, 成为速率匹配模块的输入 比特序列, 经速率匹配输出同物理信道的无线帧的比特数相匹配的输出比特序列, 当对 速率匹配的打孔有最大打孔比例限制时, 那些会使打孔数超过该比例限制的发送数据会 被主动丢弃, 当不对速率匹配的打孔进行最大打孔比例限制时, 那些打孔比例过大的发 送数据在接收端也无法被正确地还原。 由于上述原因, 等价于物理信道无线帧比特数的 物理信道速率, 可决定传输信道格式组合的有效性; 有效传输格式组合集中对应于自适 应多速率(AMR) 20毫秒话音帧模式的那些传输信道格式组合, 决定了有效的自适应多 速率 (AMR)话音帧模式。  In the Universal Mobile Telecommunications System (UMTS) system, the transmission data in the transmission channel is sent to the composite channel, and is subjected to operation steps such as CRC pasting, transport block concatenation/code block splitting, channel coding, etc., and becomes an input bit sequence of the rate matching module. The output bit sequence that matches the number of bits of the radio frame of the physical channel by the rate matching output, when the rate matching puncturing has the maximum puncturing ratio limit, the transmission data that causes the number of puncturing to exceed the ratio limit will be Active discarding. When the maximum punch ratio is not limited for rate matching punches, those transmitted data with too large punch ratio cannot be correctly restored at the receiving end. For the above reasons, the physical channel rate equivalent to the number of physical frame radio frame bits can determine the validity of the transport channel format combination; the effective transport format combination concentrates those transmissions corresponding to the adaptive multi-rate (AMR) 20 millisecond voice frame mode. The combination of channel formats determines the effective adaptive multi-rate (AMR) voice frame mode.
通用移动通信*** (UMTS ) 引入了传输信道和传输格式组合, 在周期性时间间隔 内从逻辑信道调度数据到传输信道, 该时间间隔称为传输时间间隔 (TTI), 数据是按照 传输格式(TF)在给定的传输时间间隔(ΤΉ)间隔期间被发送的, 该传输格式还固定了 要被发送数据块的大小、 数目、 纠错编码方式和速率匹配参数。 用于构建复合信道的传 输信道的传输格式列表构成了传输格式组合 (TFC)。 无线资源控制 (RRC)单元为媒体介入 控制单元 MAC和物理层配置传输格式组合集合 (TFCS)。 选择对于每个传输信道适当的 传输格式,在相应的传送时间间隔 (ΤΉ)期间内在相关的传输信道内发送逻辑信道的数据, 对传输信道选择必须使传输格式的组合 (TFC)属于传输格式组合集合 (TFCS)。在有关通用 移动通信***(UMTS)标准的说明书中通常将以上操作称为 "传输格式组合(TFC)选 择"。  The Universal Mobile Telecommunications System (UMTS) introduces a combination of transport channel and transport format to schedule data from a logical channel to a transport channel during a periodic time interval. This time interval is called the transmission time interval (TTI) and the data is in accordance with the transport format (TF). The transmission format is fixed during a given transmission time interval (ΤΉ) interval, which also fixes the size, number, error correction coding mode, and rate matching parameter of the data block to be transmitted. The transport format list of the transport channels used to construct the composite channel constitutes a Transport Format Combination (TFC). The Radio Resource Control (RRC) unit configures a Transport Format Combination Set (TFCS) for the Media Intervention Control Unit MAC and Physical Layer. Selecting the appropriate transport format for each transport channel, transmitting the data of the logical channel within the relevant transport channel during the corresponding transmission time interval (ΤΉ), the transport channel selection must be such that the transport format combination (TFC) belongs to the transport format combination Collection (TFCS). The above operation is generally referred to as "Transport Format Combination (TFC) Selection" in the specification of the Universal Mobile Telecommunications System (UMTS) standard.
有效自适应多速率 (AMR)编码帧是 3个或 2个或 1个类 (class) 的比特, 每类的 比特占用一个传输信道。 现有技术使用的是固定的 20毫秒传输时间间隔 (ΤΉ) 的传输 格式组合, 以用该组合中的一个传输格式调度一个自适应多速率 (AMR)帧的一个类的 比特到该格式的传输信道上的方式发送自适应多速率 (AMR)帧,将自适应多速率 (AMR) 编码器每 20毫秒输出的 1个帧由 20毫秒传输时间间隔(TTI) 内的传输块来承载, 在每 次传输格式组合选择时根据自适应多速率 (AMR)编码器输出的编码帧的模式选定每个 类对应的传输信道的传输格式, 话音数据的传输信道的格式和其它数据的传输信道的格 式一起构成了传输格式组合, 当然该传输格式组合是传输格式组合集合中的有效传输格 式组合。 A valid adaptive multi-rate (AMR) coded frame is three or two or one class of bits, each class occupying one transmission channel. The prior art uses a fixed 20 millisecond transmission time interval (ΤΉ) transport format combination to schedule a class of an adaptive multi-rate (AMR) frame to a transmission of the format using one of the combinations. The method on the channel transmits an adaptive multi-rate (AMR) frame, and one frame outputted by the adaptive multi-rate (AMR) encoder every 20 milliseconds is carried by a transport block within a 20 millisecond transmission time interval (TTI), at each When the secondary transmission format combination is selected, the transmission format of the transmission channel corresponding to each class is selected according to the mode of the encoded frame output by the adaptive multi-rate (AMR) encoder, the format of the transmission channel of the voice data, and the format of the transmission channel of other data. Together, the transport format combination is formed, which of course is a combination of valid transport formats in the transport format combination set.
当移动台在进行通用移动通信*** (UMTS ) 的切换时, 移动台和无线接入网之间 的无线链路发生改变: 硬切换时这种改变是无线链路的暂时中断; 软切换是无间断的切 换, 但在切换期间往往有暂时的无线链路物理信道速率的下降。 由此可见, 在切换期间 因为这种物理信道的变化会引起若干自适应多速率(AMR) 20毫秒话音帧的丢失。  When the mobile station is performing a Universal Mobile Telecommunications System (UMTS) handover, the radio link between the mobile station and the radio access network changes: this change is a temporary interruption of the radio link during hard handover; the soft handover is none Intermittent switching, but there is often a temporary decrease in the physical channel rate of the radio link during handover. It can be seen that this change in physical channel causes a loss of several adaptive multi-rate (AMR) 20 millisecond voice frames during handover.
话音的实时性要求话音帧以尽可能快的速度被传送和接收,移动台每 20毫秒从无线 接入协议功能单元获取一个自适应多速率 (AMR)话音帧来译码, 当无线接入协议功能 单元无法提供或提供的话音帧中有错误时, 自适应多速率 (AMR) 能以差错隐藏技术避 免少量话音帧丢失所带来的负面效应。  The real-time nature of the voice requires that the voice frame be transmitted and received as fast as possible, and the mobile station acquires an adaptive multi-rate (AMR) voice frame from the wireless access protocol functional unit every 20 milliseconds to decode, when the wireless access protocol When there is an error in the voice frame that the functional unit cannot provide or provide, Adaptive Multi-Rate (AMR) can avoid the negative effects of a small number of voice frame loss with error concealment techniques.
发送含自适应多速率 (AMR)话音帧的一方是以每 20毫秒调度一帧的固定速率发 送, 当在某个 20毫秒有一帧无法发送时丢弃该帧, 在下个 20毫秒调度下一个话音帧, 连续的每 20毫秒一个自适应多速率 (AMR)话音帧来自于自适应多速率(AMR) 话音 编码器对话音的编码。 对于每个 20毫秒的话音来说, 它的自适应多速率(AMR) 话音 帧的空中发送时间是固定的、 不可变动的。  The party transmitting the adaptive multi-rate (AMR) voice frame is sent at a fixed rate of one frame every 20 milliseconds. When a frame cannot be transmitted in a certain 20 milliseconds, the frame is discarded, and a voice frame is scheduled in the next 20 milliseconds. An adaptive multi-rate (AMR) speech frame every 20 milliseconds from the encoding of an adaptive multi-rate (AMR) speech encoder speech. For each 20 millisecond voice, the air transmission time of its adaptive multi-rate (AMR) voice frame is fixed and unchangeable.
自适应多速率(AMR)的概念为移动台(UE)和网络之间的连接提供了一种多速率 能力,这种能力反映在输出到逻辑信道的话音数据的自适应多速率(AMR)编码模式上, 当前技术规范并没有描述利用自适应多速率 (AMR)概念优化传输控制的极好方式和所 能允许的话音帧丢失的限制, 所以说, 通用移动通信*** (UMTS) ***的当前技术规 范为促进自适应多速率 (AMR)概念的应用留出了空间。 发明内容  The concept of adaptive multi-rate (AMR) provides a multi-rate capability for the connection between a mobile station (UE) and the network, which is reflected in adaptive multi-rate (AMR) coding of voice data output to the logical channel. In terms of mode, the current technical specifications do not describe the excellent way to optimize transmission control using the adaptive multi-rate (AMR) concept and the allowable loss of voice frame loss, so the current technology of the Universal Mobile Telecommunications System (UMTS) system. The specification leaves room for the application of the adaptive multi-rate (AMR) concept. Summary of the invention
本发明要解决的技术问题是减少话音呼叫过程中的自适应多速率(AMR) 20毫秒话 音帧的丢失, 尽管自适应多速率(AMR) 能以差错隐藏技术来避免话音帧丢失所带来的 负面效应, 但差错隐藏技术的这种避免作用只是对个别帧的丢失有好的效果, 因为它用 已收到的相邻话音帧去构建丢失帧, 所以构建的丢失帧与实际丢失帧之间有误差, 当丢 失帧之间间隔很小甚至是连续时, 后面的丢失帧的误差会变得很大。  The technical problem to be solved by the present invention is to reduce the loss of an adaptive multi-rate (AMR) 20-millisecond speech frame during a voice call, although adaptive multi-rate (AMR) can avoid error frame loss by error concealment techniques. Negative effects, but this avoidance of error concealment techniques only works well for the loss of individual frames, because it uses the adjacent voice frames that have been received to construct the lost frames, so between the constructed lost frames and the actual lost frames. There is an error, and when the interval between lost frames is small or even continuous, the error of the subsequent lost frames becomes large.
自适应多速率 (AMR)话音帧的丢失在下列三个地方比较突出, 一是物理信道速率 的突然下降引起的话音帧的丢失, 二是优先调度突发的高优先级逻辑信道所引起的话音 帧的丢失, 三是切换期间的话音帧的丢失。  The loss of adaptive multi-rate (AMR) voice frames is prominent in the following three places: one is the loss of voice frames caused by the sudden drop of the physical channel rate, and the other is the voice caused by the priority scheduling of the burst high-priority logical channels. The loss of the frame, and the third is the loss of the voice frame during the handover.
物理信道速率的突然下降可以使原来有效的自适应多速率 (AMR)话音帧传输格式 组合变成无效, 从而引起自适应多速率 (AMR) 话音帧丢失, 要适应这种变化, 一是要 改变自适应多速率 (AMR)话音编码模式, 一般需要 40毫秒以上的等待时间, 这是因 为移动台从收到的无线帧中获取物理信道速率改变的信息要一个 20毫秒无线帧的时间, 改变自适应多速率 (AMR)话音编码模式至少要等待 20毫秒的时间才能生效; 二是要 使用新的传输格式组合 (TFC), 当它是需要配置的传输格式组合 (TFC) 时, 需要一个 信令配置过程。 A sudden drop in the physical channel rate can invalidate the original effective adaptive multi-rate (AMR) voice frame transmission format combination, causing adaptive multi-rate (AMR) voice frame loss. To accommodate this change, one must change The adaptive multi-rate (AMR) voice coding mode generally requires a waiting time of 40 milliseconds or more, because the mobile station acquires the information of the physical channel rate change from the received radio frame to a time of 20 milliseconds of radio frame. Changing the Adaptive Multi-Rate (AMR) voice coding mode takes at least 20 milliseconds to take effect; the second is to use the new Transport Format Combination (TFC), which is required when it is a Transport Format Combination (TFC) that needs to be configured. Signaling configuration process.
当有突发的高优先级逻辑信道需要调度时, 在进行传输格式组合 (TFC)选择时, 原来传送自适应多速率 (AMR)话音帧传输格式组合不再被选择, 从而引起自适应多速 率(AMR)话音帧丢失, 也就是偷帧。 在 UMTS***中非话音业务比重增大, 并且信令 消息的复杂度和数量均有较大的提高, 更重要的是在 UMTS***中使用映射到物理信道 的复合信道来承载所有业务的逻辑信道, 不像 GPRS将话音业务的物理信道同分组业务 的物理信道分开, 使得, 当分组业务信道的调度抢占了物理信道时, 尤其当分组业务的 逻辑信道的优先级高于话音逻辑信道时, 话音信道被偷帧的次数大大增加, 需要有减少 偷帧的负面影响的机制。  When a bursty high-priority logical channel needs to be scheduled, when transmitting a transport format combination (TFC) selection, the original transport adaptive multi-rate (AMR) voice frame transport format combination is no longer selected, resulting in an adaptive multi-rate. (AMR) The voice frame is lost, that is, the frame is stolen. In the UMTS system, the proportion of non-voice services increases, and the complexity and number of signaling messages are greatly improved. More importantly, the composite channel mapped to the physical channel is used in the UMTS system to carry logical channels of all services. Unlike GPRS, the physical channel of the voice service is separated from the physical channel of the packet service, so that when the scheduling of the packet traffic channel preempts the physical channel, especially when the logical channel of the packet service has a higher priority than the voice logical channel, the voice The number of times a channel is stolen is greatly increased, and a mechanism for reducing the negative impact of stealing frames is required.
切换期间,如果是硬切换就有无线链路的暂时中断, 必然引起自适应多速率(AMR) 话音帧丢失; 如果是软切换且有暂时的无线链路物理信道速率的下降, 也可能引起自适 应多速率 (AMR) 话音帧丢失。  During the handover, if there is a temporary interruption of the radio link due to the hard handover, the adaptive multi-rate (AMR) voice frame will be lost. If it is a soft handover and there is a temporary decrease in the physical link rate of the radio link, it may also cause Adapt to multi-rate (AMR) voice frame loss.
对于移动台来说, 如背景中所述的每次传输格式组合选择所针对的自适应多速率 (AMR) 编码帧来自于自适应多速率 (AMR) 编码器, 即, 自适应多速率 (AMR) 编 码器每 20毫秒用生成的自适应多速率(AMR)编码帧刷新一次放置自适应多速率(AMR) 编码帧的存储单元,信道编码器每 20毫秒从该存储单元读取其中的自适应多速率 (AMR) 编码帧并进行编码和复用。 而传输格式组合选择要受限于执行选择时的传送格式组合集 合 (TFCS)中的有效的那些传送格式组合, 往往由于信道的快速变化、 突发的信令消息和 ". 切换使得有效传送格式组合集合中没有能够传送从上述存储单元中实时读取的自适应多 速率(AMR)话音编码帧的有效传送格式组合, 从而使自适应多速率(AMR)编码帧丢 失。 上述现象发生的原因在于: 针对瞬间的当前无线环境和通用移动通信***(UMTS) 的 8种自适应多速率 (AMR)模式所选的最佳传输信道格式组合 (TFC)与当前帧的自 适应多速率 (AMR) 模式不一致, 调整自适应多速率 (AMR) 模式需要一定的时延。  For mobile stations, the adaptive multi-rate (AMR) coded frame for each transport format combination selection as described in the background comes from an adaptive multi-rate (AMR) encoder, ie, adaptive multi-rate (AMR) The encoder refreshes the storage unit of the adaptive multi-rate (AMR) coded frame once every 20 milliseconds with the generated adaptive multi-rate (AMR) coded frame, and the channel encoder reads the adaptive from the memory unit every 20 milliseconds. Multi-rate (AMR) encodes frames and encodes and multiplexes them. The transport format combination selection is limited by those transport format combinations that are valid in the Transport Format Combination Set (TFCS) when performing the selection, often due to rapid channel changes, bursty signaling messages, and ". There is no effective transport format combination in the combined set that is capable of transmitting adaptive multi-rate (AMR) speech encoded frames read in real time from the above described storage units, thereby causing adaptive multi-rate (AMR) encoded frames to be lost. : Optimal Transport Channel Format Combination (TFC) and adaptive multi-rate (AMR) mode for the current frame selected for the instantaneous current wireless environment and the eight adaptive multi-rate (AMR) modes of the Universal Mobile Telecommunications System (UMTS) Inconsistent, adjusting the adaptive multi-rate (AMR) mode requires a certain amount of delay.
本发明为了解决上述技术问题, 对自适应多速率 (AMR) 话音在编码控制、 传输信 道上的发送接收、 以及核心网 -无线接入网 (Iu)接口上发送接收方面给出了技术方案。  In order to solve the above technical problem, the present invention provides a technical solution for adaptive multi-rate (AMR) voice transmission control on a coding control, a transmission channel, and transmission and reception on a core network-radio access network (Iu) interface.
——移动台中生成和传送自适应多速率 (AMR)编码帧的方法。  - A method of generating and transmitting adaptive multi-rate (AMR) coded frames in a mobile station.
本发明的在移动台中的编码和发送的方案是一种在移动台中生成和传送自适应多速 率(AMR)编码帧的方法, 在话音呼叫过程中, 自适应多速率(AMR)编码器通过对输 入话音信号的采样得到长度为 20毫秒的话音帧,每个话音帧含 20毫秒话音的采样样本, 对 20毫秒话音帧编码, 所述编码产生的是符合通用移动通信*** (UMTS)标准的自适 应多速率(AMR)编码帧, 需要无线接口功能单元发送自适应多速率(AMR)编码帧形 式的话音数据, 无线接口功能单元按各传输信道的传输格式发送所述的话音数据, 其特 征在于: The scheme of encoding and transmitting in a mobile station of the present invention is a method of generating and transmitting an adaptive multi-rate (AMR) coded frame in a mobile station. During a voice call, an adaptive multi-rate (AMR) encoder passes The input voice signal is sampled to obtain a voice frame of 20 milliseconds in length. Each voice frame contains a sample sample of 20 millisecond voice, and encodes a 20 millisecond voice frame, which is generated in accordance with the Universal Mobile Telecommunications System (UMTS) standard. To adapt to a multi-rate (AMR) coded frame, the wireless interface function unit is required to transmit voice data in the form of an adaptive multi-rate (AMR) coded frame, and the wireless interface function unit transmits the voice data according to the transmission format of each transport channel. The levy is:
向自适应多速率(AMR) 编码器发出编码命令, 所述编码命令指定了多个自适应多 速率(AMR)模式, 自适应多速率 (AMR)编码器根据所述编码命令对一个 20毫秒话 音帧编码所产生的有效话音帧编码序列, 要么是多个自适应多速率 (AMR)话音编码帧 的集合, 要么是一个自适应多速率 (AMR)静默编码帧, 当所述话音帧编码序列是多个 自适应多速率(AMR)话音编码帧时, 其自适应多速率(AMR)编码帧的模式同所述编 码命令中的自适应多速率(AMR)模式相一致;  An encoding command is issued to an adaptive multi-rate (AMR) encoder, the encoding command specifying a plurality of adaptive multi-rate (AMR) modes, the adaptive multi-rate (AMR) encoder according to the encoding command to a 20 millisecond voice The effective voice frame coding sequence generated by frame coding is either a set of multiple adaptive multi-rate (AMR) voice coded frames or an adaptive multi-rate (AMR) silence coded frame, when the voice frame coding sequence is When multiple adaptive multi-rate (AMR) speech encoded frames, the mode of the adaptive multi-rate (AMR) encoded frame is consistent with the adaptive multi-rate (AMR) mode in the encoded command;
向无线接口功能单元输出话音帧编码序列;  Outputting a voice frame coding sequence to the wireless interface function unit;
在无线接口功能单元中, 发送一个话音帧编码序列中的自适应多速率(AMR) 编码 帧到传输信道的过程, 包括, 在该话音帧编码序列中挑选一个自适应多速率 (AMR) 编 码帧和挑选一个传输格式组合, 挑选出的传输格式组合, 包含了传送挑选出的自适应多 速率 (AMR) 编码帧所有类比特的传输格式, 用该传输格式组合将挑选出的自适应多速 率(AMR)编码帧调度到传输信道上。 在选定一个自适应多速率(AMR)编码帧的发送 后丢弃话音帧编码序列中其余的自适应多速率(AMR) 编码帧。  In the wireless interface functional unit, the process of transmitting an adaptive multi-rate (AMR) coded frame in a voice frame coding sequence to a transmission channel, comprising selecting an adaptive multi-rate (AMR) coded frame in the voice frame coding sequence And selecting a transport format combination, the selected transport format combination, including the transmission format of all the bits of the selected adaptive multi-rate (AMR) coded frame, and using the transport format combination to select the adaptive multi-rate ( AMR) The coded frame is scheduled onto the transport channel. The remaining adaptive multi-rate (AMR) coded frames in the voice frame coding sequence are discarded after transmission of an adaptive multi-rate (AMR) coded frame is selected.
上述在移动台中编码控制和发送的方案中的无线接口功能单元是为移动台的用户数 据提供空中接口的功能单元, 它的输入是话音、 分组和信令这些用户数据, 输出的是无 线帧,它要符合无线资源控制(RRC)、无线链路控制(RLC)、媒体接入控制单元(MAC) 和物理层 (PHY) 的通信协议。  The above-mentioned wireless interface functional unit in the scheme of coding control and transmission in the mobile station is a functional unit that provides an air interface for the user data of the mobile station, and its input is user data such as voice, packet and signaling, and the output is a radio frame. It is compliant with communication protocols for Radio Resource Control (RRC), Radio Link Control (RLC), Media Access Control Unit (MAC), and Physical Layer (PHY).
使用上述的在移动台中编码控制和发送的方案, 当话音逻辑信道上发送的是非静默 帧时: 如果与前一次调度话音逻辑信道到传送信道时传送非静默帧时的传输格式组合 Using the above scheme of encoding control and transmission in the mobile station, when a non-silent frame is transmitted on the voice logical channel: If the transmission format combination is transmitted when the non-silent frame is transmitted from the previous scheduled logical channel to the transmission channel
(TFC) 相比, 当需要被调度到传送信道上去的高优先级非话音发送逻辑信道发生改变 时, 或者当物理信道的最高传输速率发生改变引起有效传输格式组合(TFC)集合变化 时, 调度话音逻辑信道到传送信道时的传输格式组合 (TFC)选择不再是前一次在传输 信道上传送的自适应多速率 (AMR) 非静默编码帧的模式挑选合适的传输格式组合(TFC), when a high-priority non-voice transmission logical channel that needs to be scheduled on the transport channel changes, or when the highest transmission rate of the physical channel changes, causing a change in the effective transport format combination (TFC) set, The transport format combination (TFC) selection of the voice logical channel to the transport channel is no longer the mode of adaptive multi-rate (AMR) non-silently coded frames previously transmitted on the transport channel.
(TFC), 这是因为当前有多种自适应多速率(AMR)非静默编码帧的模式可供挑选。这 样带来的好处是: 由于一次编码产生多种模式的自适应多速率 (AMR)非静默编码帧, 当无法为前一次的模式挑选到传输格式组合 (TFC ) 时, 可能找到另一自适应多速率(TFC), because there are currently a number of adaptive multi-rate (AMR) non-silently encoded frames available for selection. The benefits of this are: Since one-time encoding produces multiple modes of adaptive multi-rate (AMR) non-silently encoded frames, another adaptation may be found when the transport format combination (TFC) cannot be selected for the previous mode. Multiple rate
(AMR)非静默编码帧模式的传输格式组合(TFC); 当有对应多种模式的多种传输格式 组合(TFC)可供选择时,可选用比特率最高的模式的传输格式组合(TFC)去传送话音。 (AMR) Transport Format Combination (TFC) for non-silently coded frame mode; when there are multiple Transport Format Combinations (TFC) available for multiple modes, the Transport Format Combination (TFC) of the mode with the highest bit rate is available. To transmit voice.
为了解决上述找不到有效的传送格式组合来将从上述存储单元中实时读取的自适应 多速率 (AMR) 话音编码帧调度到传输信道上这个问题, 在本发明的移动台中, 先进先 出 (FIFO)缓存替代了原来的存储单元接受自适应多速率(AMR)编码器输出的话音帧 编码序列。  In order to solve the above problem that the above-mentioned effective transport format combination cannot be found to schedule an adaptive multi-rate (AMR) voice coded frame read from the above-mentioned storage unit to the transmission channel, in the mobile station of the present invention, the first in first out The (FIFO) buffer replaces the voice frame coding sequence that the original memory unit accepts from the adaptive multi-rate (AMR) encoder output.
本发明提供一种通用移动通信***中的移动台, 该移动台的自适应多速率(AMR) 编码器对 20毫秒长度话音帧编码产生话音帧编码序列的输出,并且该移动台包括一与自 适应多速率(AMR)编码器输出端相连接的先进先出(FIFO)缓存, 该先进先出(FIFO) 缓存包括- 一数据输入接口, 用于读入自适应多速率 (AMR) 编码器编码产生的话音帧编码序 列, The present invention provides a mobile station in a universal mobile communication system, the mobile station's adaptive multi-rate (AMR) The encoder encodes a 20 millisecond length voice frame encoding to produce an output of the voice frame encoding sequence, and the mobile station includes a first in first out (FIFO) buffer coupled to the output of the adaptive multi-rate (AMR) encoder, the first in first out The (FIFO) buffer includes - a data input interface for reading a sequence of voice frame codes generated by adaptive multi-rate (AMR) encoder encoding,
一数据输出接口, 用于从该先进先出 (FIFO) 缓存读取被存储的话音帧编码序列, 一存储状态接口, 用于输出该先进先出 (FIFO) 缓存中存储的话音帧编码序列的数 量和其中的话音帧编码序列的类型。  a data output interface for reading a stored voice frame code sequence from the first in first out (FIFO) buffer, a memory state interface for outputting a voice frame code sequence stored in the first in first out (FIFO) buffer The number and type of voice frame encoding sequence in it.
先进先出(FIFO)缓存的存放限制为一个话音帧编码序列时, 当 20毫秒内无法发送 话音时, 即 20毫秒内没有进行读取操作时, 该话音帧编码序列会被后续的写入所覆盖。 为了避免被覆盖的现象, 可以在上述的先进先出 (FIFO) 缓存存储 2个以上的话音帧编 码序列, 便于在未能读取的 20毫秒后有机会读取, gp, 采用后面所述的移动台中以间断 方式生成和传送自适应多速率 (AMR) 编码帧的方法。  When the storage of the first-in-first-out (FIFO) buffer is limited to a voice frame coding sequence, when the voice cannot be transmitted within 20 milliseconds, that is, when the read operation is not performed within 20 milliseconds, the voice frame coding sequence is subsequently written. cover. In order to avoid the phenomenon of being covered, more than two voice frame coding sequences can be stored in the above-mentioned first-in, first-out (FIFO) buffer, so that it is easy to read after 20 milliseconds of failure to read, gp, using the following A method of generating and transmitting adaptive multi-rate (AMR) coded frames in a discontinuous manner in a mobile station.
移动台从所述先进先出 (FIFO) 缓存的存储状态接口读取待处理的话音帧编码序列 的类型, 当待处理的话音帧编码序列是需要发送的有效话音帧编码序列时, 为该话音帧 编码序列选择传输格式组合, 用所选择的传输格式组合传输该话音帧编码序列中的一个 自适应多速率 (AMR)编码帧。  The mobile station reads the type of the voice frame coding sequence to be processed from the storage state interface of the first in first out (FIFO) buffer, and when the voice frame coding sequence to be processed is a valid voice frame coding sequence to be transmitted, the voice is The frame coding sequence selects a transport format combination, and transmits an adaptive multi-rate (AMR) coded frame in the voice frame coding sequence in combination with the selected transmission format.
根据移动台的无线接口协议功能单元配置的传输格式 (TF) 可确定它所对应的自适 应多速率 (AMR)模式编码帧的类(class) 比特, 再根据配置的传输格式组合 (TFC) 确定它所对应的自适应多速率(AMR)模式,这样,就可从配置的传输格式组合集(TFCS) 获得编码命令中的模式的可选范围。  The transport format (TF) configured according to the radio interface protocol functional unit of the mobile station can determine the class bits of the adaptive multi-rate (AMR) mode encoded frame corresponding thereto, and then determine according to the configured transport format combination (TFC). It corresponds to an adaptive multi-rate (AMR) mode so that an optional range of modes in the encoded command can be obtained from the configured Transport Format Combination Set (TFCS).
在移动台中根据无线接口协议来确定编码命令中模式的编码控制和发送的方案是: 按照上述的在移动台中生成和传送自适应多速率 (AMR) 编码帧的方法, 所述编码命令 中的每个自适应多速率(AMR)模式同移动台的传输格式组合集中的一个话音传输格式 组合相对应, 所述话音传输格式组合是包含一种自适应多速率(AMR)模式编码帧所有 类比特传输格式的传输格式组合。  A scheme for determining encoding control and transmission of a mode in an encoding command according to a radio interface protocol in a mobile station is: a method of generating and transmitting an adaptive multi-rate (AMR) encoded frame in a mobile station as described above, each of said encoding commands The adaptive multi-rate (AMR) mode corresponds to a voice transmission format combination of the mobile station's transport format combination set, which includes an adaptive multi-rate (AMR) mode coded frame for all types of bit transmissions. The format of the transport format combination.
这样做带来的好处是: 移动台可以使自适应多速率 (AMR)编码器输出的帧的模式 同变化的传输格式组合(TFC) 相匹配, 移动台无线接口协议功能单元能及时地将无线 链路上的变化反映给自适应多速率 (AMR)编码器, 特别是无线链路处于快速变化状态 时, 例如高速运动时, 无线接口协议功能单元能检测出这种快速变化状态, 就可要求编 码器以输出多模式的自适应多速率 (AMR)编码帧的方式来适应这种快速变化。  The benefits of this are: The mobile station can match the mode of the frame output by the adaptive multi-rate (AMR) encoder with the changed transport format combination (TFC), and the mobile station wireless interface protocol functional unit can wirelessly The changes on the link are reflected to the adaptive multi-rate (AMR) encoder. Especially when the wireless link is in a fast changing state, such as high-speed motion, the wireless interface protocol function unit can detect such a fast change state and can request The encoder adapts to this fast change in a manner that outputs multi-mode adaptive multi-rate (AMR) encoded frames.
——网络侧的自适应多速率 (AMR)编码和发送的方法。  - Adaptive multi-rate (AMR) coding and transmission methods on the network side.
在通用移动通信*** (UMTS) 的网络侧的编码和发送的方案同在移动台中的相类 似, 本发明给出在通用移动通信*** (UMTS) 的网络侧的一种生成和传送自适应多速 率 (AMR)编码帧的方法, 在话音呼叫过程中, 在核心网中的码型变换器将输入的话音 编码信号转换成自适应多速率(AMR)编码帧, 所述自适应多速率(AMR)编码帧被放 置到 Iu用户平面帧上向无线网络控制器(RNC)发送, 无线网络控制器(R C)在传输 信道上发送话音数据, 其特征在于- 向码型变换器发送编码命令,所述编码命令指定了多个自适应多速率(AMR)模式, 码型变换器按所述编码命令, 对输入的话音编码信号进行编码操作, 所产生的有效的 20 毫秒长度话音的话音帧编码序列, 要么是若干 (大于等于 1 )个自适应多速率 (AMR) 话音编码帧的集合, 要么是一个自适应多速率 (AMR) 静默编码帧, 当所述话音帧编码 序列是多 (大于 1 )个自适应多速率 (AMR)话音编码帧时, 其自适应多速率 (AMR) 编码帧的模式同所述编码命令中的自适应多速率 (AMR)模式相一致; The coding and transmission scheme on the network side of the Universal Mobile Telecommunications System (UMTS) is similar to that in the mobile station, and the present invention provides an adaptive multi-speed generation and transmission on the network side of the Universal Mobile Telecommunications System (UMTS). Rate (AMR) method of encoding frames, during a voice call, a pattern converter in the core network converts the input voice encoded signal into an adaptive multi-rate (AMR) encoded frame, said adaptive multi-rate (AMR) The coded frame is placed on the Iu user plane frame and sent to the Radio Network Controller (RNC), and the Radio Network Controller (RC) transmits voice data on the transport channel, characterized by - transmitting an encoding command to the pattern converter, The encoding command specifies a plurality of adaptive multi-rate (AMR) modes, and the pattern converter encodes the input voice encoded signal according to the encoding command, and generates a valid voice frame encoding sequence of 20 milliseconds length voice. , either a set of several (greater than or equal to 1) adaptive multi-rate (AMR) voice coded frames, or an adaptive multi-rate (AMR) silence coded frame, when the voice frame coding sequence is multiple (greater than 1) An adaptive multi-rate (AMR) voice coded frame whose mode of adaptive multi-rate (AMR) coded frame is consistent with an adaptive multi-rate (AMR) mode in the coded command;
用核心网-无线接入网接口 (Iu) 用户平面帧承载话音帧编码序列中的所有自适应多 速率 (AMR)编码帧, 向无线网络控制器(R C) 发送所述承载话音帧编码序列的核心 网-无线接入网接口 (Iu)用户平面帧;  Transmitting, by the core network-radio access network interface (Iu) user plane frame, all adaptive multi-rate (AMR) coded frames in the voice frame coding sequence, transmitting the bearer voice frame coding sequence to the radio network controller (RC) Core network - radio access network interface (Iu) user plane frame;
在无线网络控制器(RNC)中, 调度一个话音帧编码序列中的自适应多速率(AMR) 编码帧到传输信道的过程,包括,在该话音帧编码序列中挑选一个自适应多速率(AMR) 编码帧和挑选一个传输格式组合, 挑选出的传输格式组合, 包含了传送挑选出的自适应 多速率 (AMR)编码帧所有类比特的传输格式, 用该传输格式组合将挑选出的自适应多 速率(AMR)编码帧调度到传输信道上。在选定一个自适应多速率(AMR)编码帧的发 送后丢弃话音帧编码序列中其余的自适应多速率 (AMR)编码帧。  In a Radio Network Controller (RNC), the process of scheduling an adaptive multi-rate (AMR) coded frame in a voice frame coding sequence to a transmission channel, including selecting an adaptive multi-rate (AMR) in the sequence of voice frame codes Encoding the frame and selecting a combination of transmission formats, and selecting the combination of the transmission formats, including the transmission format of all the bits of the selected adaptive multi-rate (AMR) coded frame, and using the combination of the transmission formats to select the adaptive Multi-rate (AMR) coded frames are scheduled onto the transport channel. The remaining adaptive multi-rate (AMR) coded frames in the voice frame coding sequence are discarded after transmission of an adaptive multi-rate (AMR) coded frame is selected.
在通用移动通信*** (UMTS) 网络侧由无线网络控制器(RNC)确定编码命令中 模式的编码生成和发送的方案是, 按照上述的通用移动通信*** (UMTS ) 网络侧中的 生成和传送自适应多速率 (AMR) 编码帧的方法, 无线网络控制器(R C) 向核心网发 送核心网-无线接入网接口 (Iu)用户平面速率控制帧, 核心网-无线接入网接口 (Iu)用 户平面速率控制帧中的多个无线接入承载子流组合指示的序号标识(RFCI n indicator) 域与多个无线接入承载子流组合指示(RFCI) —一对应, 这些无线接入承载子流组合指 示 (RPCI)又与自适应多速率 (AMR)模式相对应, 核心网收到所述核心网-无线接入 网接口 (Iu)用户平面速率控制帧后, 向码型变换器发送的编码命令, 编码命令中指定 的多个自适应多速率(AMR)模式就是所述无线接入承载子流组合指示(RFCI)所对应 的自适应多速率(AMR) 话音模式。  The scheme for determining the encoding generation and transmission of the mode in the encoding command by the radio network controller (RNC) on the network side of the Universal Mobile Telecommunications System (UMTS) is generated and transmitted in the network side of the Universal Mobile Telecommunications System (UMTS) as described above. A method for adapting a multi-rate (AMR) coded frame, the radio network controller (RC) transmits a core network-radio access network interface (Iu) user plane rate control frame to the core network, and a core network-radio access network interface (Iu) The radio frequency identification indicator (RFCI n indicator) field of the plurality of radio access bearer substream combination indications in the user plane rate control frame is corresponding to a plurality of radio access bearer substream combination indications (RFCI), and the radio access bearers are The stream combination indication (RPCI), in turn, corresponds to an adaptive multi-rate (AMR) mode, and after the core network receives the core network-radio access network interface (Iu) user plane rate control frame, sends the signal to the pattern converter. Encoding command, the multiple adaptive multi-rate (AMR) modes specified in the coding command are the multiple adaptations corresponding to the radio access bearer substream combination indication (RFCI) Rate (AMR) voice mode.
——延迟发送和译码的原理和根据。  - Principle and basis for delayed transmission and decoding.
为避免话音帧丢失, 本发明提出了将传输信道或物理信道停止传送话音数据期间的 自适应多速率(AMR)编码帧延迟发送的方法, 以及相应的移动台自适应多速率(AMR) 延迟译码的方法。 延迟发送要结合上述生成和发送话音帧编码序列的方法, 因为延迟后 再次发送时的信道和原来准备发送时的信道可能区别较大, 不能沿用原先的自适应多速 率 (AMR)话音的模式, 必须寻找合适的自适应多速率 (AMR) 的模式。 In order to avoid loss of voice frames, the present invention proposes a method for delay transmission of adaptive multi-rate (AMR) coded frames during transmission of voice channels or physical channels to stop transmitting voice data, and corresponding mobile station adaptive multi-rate (AMR) delay translation. The method of the code. Delayed transmission is combined with the above-mentioned method of generating and transmitting a voice frame coding sequence, because the channel when transmitting again after delay may be different from the channel originally intended to be transmitted, and the original adaptive multi-speed cannot be used. The rate (AMR) voice mode must look for a suitable adaptive multi-rate (AMR) mode.
实时话音的特性是: 端到端时延小, 最大允许端到端时延要根据人对音频会话的感 觉来决定, 一般主观评估端到端时延在 200-300毫秒之间是可接受的, 因此只要本发明 的方法把端到端时延控制到 200毫秒以下是不会引起明显的时延方面的业务质量下降的。  The characteristics of real-time voice are: The end-to-end delay is small, and the maximum allowable end-to-end delay is determined according to the person's perception of the audio session. Generally, the subjective evaluation of the end-to-end delay is acceptable between 200-300 milliseconds. Therefore, as long as the method of the present invention controls the end-to-end delay to less than 200 milliseconds, it does not cause significant degradation in service quality in terms of delay.
以下先说明移动台延迟自适应多速率 (AMR) 译码的方法, 然后, 分别对移动台和 网络侧的缓存式的发送方法做出说明。  The following describes the method of mobile station delay adaptive multi-rate (AMR) decoding, and then describes the method of buffered transmission on the mobile station and the network side, respectively.
——移动台的延迟译码的方法。  - A method of delay decoding for mobile stations.
本发明提出的移动台自适应多速率(AMR)译码器延迟译码的方案是: 一种移动台 的处理接收到的自适应多速率 (AMR)编码帧的方法, 移动台从传送话音的传输信道上 接收自适应多速率(AMR)编码帧, 自适应多速率(AMR)译码器对移动台收到的自适 应多速率 (AMR)编码帧按从先到后的次序每 20毫秒完成一个自适应多速率 (AMR) 编码帧的译码输出, 其特征在于- 为一个话音呼叫设立放置自适应多速率 (AMR) 编码帧的一个缓存区和该缓存区的 下限值, 从移动台第一次收到所述话音呼叫的自适应多速率 (AMR) 编码帧开始, 将收 到的自适应多速率 (AMR)话音编码帧存放到缓存区内,当缓存区内自适应多速率 (AMR) 编码帧的个数达到所述下限值时, 自适应多速率 (AMR)译码器开始对移动台收到的自 适应多速率(AMR) 编码帧译码,  The mobile station adaptive multi-rate (AMR) decoder delay decoding scheme proposed by the present invention is: a method for a mobile station to process an adaptive multi-rate (AMR) coded frame, the mobile station transmitting voice An adaptive multi-rate (AMR) coded frame is received on the transport channel, and the adaptive multi-rate (AMR) decoder performs the adaptive multi-rate (AMR) coded frame received by the mobile station in a first to last order every 20 milliseconds. A decoded output of an adaptive multi-rate (AMR) coded frame, characterized by - setting up a buffer area for placing an adaptive multi-rate (AMR) coded frame for a voice call and a lower limit value of the buffer area, from the mobile station The first time an adaptive multi-rate (AMR) coded frame of the voice call is received, the received adaptive multi-rate (AMR) voice coded frame is stored in the buffer area, and the adaptive multi-rate is in the buffer area ( AMR) When the number of coded frames reaches the lower limit, the adaptive multi-rate (AMR) decoder begins decoding the adaptive multi-rate (AMR) coded frame received by the mobile station.
当移动台处于语音(SPEECH)模式时,检査每 20毫秒缓存区内自适应多速率 (AMR) 话音编码帧增加的的数目: 当所述数目小于 1时, 主动添加自适应多速率 (AMR)接收 丢失帧; 当所述数目大于 1并且缓冲区中存在语音(SPEECH)模式下主动添加的接收丢 失帧时, 删除所述的在语音 (SPEECH)模式下主动添加的接收丢失帧。  When the mobile station is in voice (SPEECH) mode, check the number of adaptive multi-rate (AMR) voice coded frames added in the buffer area every 20 milliseconds: When the number is less than 1, actively add adaptive multi-rate (AMR) Receiving a lost frame; when the number is greater than 1 and the received lost frame is actively added in the voice (SPEECH) mode in the buffer, the received lost frame actively added in the voice (SPEECH) mode is deleted.
使用这种缓存自适应多速率 (AMR)编码帧的方法的好处在于, 它使得, 自适应多 速率 (AMR)编码帧的发送方不再如背景中所介绍的那样, 一旦无法将一个自适应多速 率 (AMR)编码帧在 20毫秒的两个无线帧中发送出去就必须丢弃此帧。 使用本发明的 方法, 即使在后续的无线帧中发送此 20毫秒话音数据中的自适应多速率 (AMR)编码 帧, 该自适应多速率 (AMR) 编码帧仍然能以正确的次序被译码, 所述的后续发送涉及 到从序列中的多种模式自适应多速率(AMR)话音编码帧里选择一种模式的帧发送, 发 送所述自适应多速率 (AMR)话音编码帧既可以是无线网络控制器(RNC) 也可以是移 动台。  The advantage of using this cache adaptive multi-rate (AMR) coded frame is that it makes the sender of the adaptive multi-rate (AMR) coded frame no longer as described in the background, once it is not possible to adapt Multi-rate (AMR) coded frames must be dropped in two radio frames of 20 milliseconds. Using the method of the present invention, the adaptive multi-rate (AMR) coded frames can be decoded in the correct order even if the adaptive multi-rate (AMR) coded frames in the 20 millisecond voice data are transmitted in subsequent radio frames. The subsequent transmission involves selecting a mode of frame transmission from a plurality of mode adaptive multi-rate (AMR) voice coded frames in the sequence, and transmitting the adaptive multi-rate (AMR) voice coded frame may be The Radio Network Controller (RNC) can also be a mobile station.
自适应多速率(AMR)编码帧使用透明模式的 RLC层模式,不对承载自适应多速率 (AMR)编码帧的协议数据单元附加序号, 接收方按照收到自适应多速率(AMR)编码 帧的时间顺序进行排序, 如果在每 20毫秒的两个无线帧中没有自适应多速率(AMR) 编码帧可供接收方使用, 接收方就***一个接收丢失帧, 接收丢失帧的类型是 3GPP TS26.101的 4丄 3的表 (Table lc)中接收类型(RX— TYPE)为无数据(NO— DATA)的类型。 由于在本发明方法中以不均匀的方式进行平均每 20毫秒一帧的自适应多速率 (AMR) 话音编码帧发送, 就需要把那些由不均匀发送引起的错误添加的接收丢失帧从缓存区删 掉, 消除由不均匀发送话音帧引起的负面效应, 这种利用缓存区存储空间适应话音传送 暂时停顿的方法使得在编码完成时的自适应多速率 (AMR)话音编码帧的互相间的时间 距离得到最大程度的保持。 The adaptive multi-rate (AMR) coded frame uses the transparent mode RLC layer mode, and does not add a sequence number to the protocol data unit carrying the adaptive multi-rate (AMR) coded frame, and the receiver follows the received adaptive multi-rate (AMR) coded frame. The chronological ordering, if there is no adaptive multi-rate (AMR) coded frame available for the receiver in two radio frames every 20 milliseconds, the receiver inserts a received lost frame, and the type of the received lost frame is 3GPP TS26. The reception type (RX_TYPE) in the 101 table of Table 101 (Table lc) is of the type of no data (NO-DATA). Since the adaptive multi-rate (AMR) voice coded frame transmission is performed every 20 milliseconds in an uneven manner in the method of the present invention, it is necessary to add the received loss frames added by the error caused by the uneven transmission from the buffer area. Deleting, eliminating the negative effects caused by unevenly transmitting voice frames, such a method of utilizing the buffer storage space to accommodate temporary pauses in voice transmission such that the time of the adaptive multi-rate (AMR) voice coded frames at the time of encoding completion The distance is kept to the maximum extent.
移动台有 2种工作模式一语音(SPEECH) 模式和舒适噪声 (COMFORT— NOISE) 模式, 在舒适噪声 (COMFORT— NOISE)模式下发送方间断地发送静默帧, 接收方主动 添加的接收丢失帧是有效的。 对于语音(SPEECH) 模式下主动添加的接收丢失帧来说, 其中部分接收丢失帧有对端多速率(AMR)话音编码器产生的并且在传送过程中丢失了 的多速率(AMR) 话音编码帧与之相对应, 另一部分则没有, 没有的这部分由本发明的 不均勾发送方式引起。那些没有对应多速率(AMR)话音编码帧的接收丢失帧是多余的, 不起补偿丢失多速率 (AMR) 话音编码帧的作用, 如果不能被不均匀发送方式的删除操 作所删除, 会引起话音的时延和质量下降, 因此需要对主动添加的接收丢失帧进行一定 限制:  The mobile station has two modes of operation-one-speech (SPEECH) mode and comfort noise (COMFORT-NOISE) mode. In the comfort noise (COMFORT-NOISE) mode, the sender intermittently transmits a silence frame, and the receiver receives the lost frame that is actively added. Effective. For received loss frames actively added in voice (SPEECH) mode, some of the received loss frames have multi-rate (AMR) voice coded frames generated by the peer multi-rate (AMR) voice coder and lost during transmission. Corresponding to this, there is no other part, and this part is not caused by the uneven sending method of the present invention. Received lost frames that do not have a corresponding multi-rate (AMR) voice-coded frame are redundant and do not compensate for the loss of multi-rate (AMR) voice-coded frames. If they cannot be deleted by the delete operation of the uneven transmission mode, the voice will be caused. The delay and quality are degraded, so there is a certain limit on the number of actively added received lost frames:
按照移动台的缓存自适应多速率(AMR)编码帧和添加和删除接收丢失帧的接收方 法, 设立所述缓存区的在语音 (SPEECH) 模式下主动添加的接收丢失帧的个数限制值, 当缓存区内语音(SPEECH)模式下主动添加的接收丢失帧的个数达到所述的个数限制值 时, 在语音(SPEECH)模式下不再添加接收丢失帧。  According to a buffer adaptive multi-rate (AMR) coded frame of the mobile station and a receiving method for adding and deleting a received lost frame, setting a limit number of the received lost frame actively added in the voice (SPEECH) mode of the buffer area, When the number of received lost frames actively added in the voice zone (SPEECH) mode reaches the limit number, the received loss frame is no longer added in the voice (SPEECH) mode.
对主动添加的接收丢失帧个数进行上述限制是因为: 在从发送移动台到接收移动台 的 RNC的路径上所能容纳的被延迟的多速率 (AMR) 编码帧的个数是有限的, 因此与 之对应的 SPEECH模式下主动添加的接收丢失帧的个数也是有限的, 应该有所述个数限 制值来体现这一有限性。  The above limitation is imposed on the number of actively added received lost frames because: the number of delayed multi-rate (AMR) coded frames that can be accommodated on the path from the transmitting mobile station to the receiving mobile station's RNC is limited, Therefore, the number of received loss frames actively added in the SPEECH mode corresponding thereto is also limited, and the number limit value should be included to reflect this limitation.
同时当发现主动添加的接收丢失帧对话音产生增加时延的影响时, 即, 当发现所述 缓存区的存储的多速率 (AMR) 编码帧个数超过了规定的缓存区上限值时, 可釆用主动 丢弃缓存区中的多速率 (AMR) 编码帧的方法来消除时延增加的负面效果。  At the same time, when it is found that the actively added received lost frame voice generates an influence of increasing delay, that is, when the number of stored multi-rate (AMR) coded frames of the buffer is found to exceed the specified upper limit of the buffer, The negative effect of increased latency can be eliminated by actively discarding multi-rate (AMR) encoded frames in the buffer.
上述方法反映到本发明提出的移动台上, 即是,一种通用移动通信***中的移动台, 该移动台包括一与自适应多速率(AMR)译码器输入端相连接的先进先出(FIFO)缓存, 该先进先出 (FIFO) 缓存包括:  The above method is reflected on the mobile station proposed by the present invention, that is, a mobile station in a universal mobile communication system, the mobile station including a first in first out connection connected to an input of an adaptive multi-rate (AMR) decoder (FIFO) cache, this first in first out (FIFO) cache includes:
一数据输入接口, 用于读入自适应多速率 (AMR)编码帧,  a data input interface for reading in an adaptive multi-rate (AMR) encoded frame,
一数据输出接口,用于从该先进先出(FIFO)缓存读出被存储的自适应多速率 (AMR) 编码帧,  a data output interface for reading the stored adaptive multi-rate (AMR) encoded frame from the first in first out (FIFO) buffer,
一存储状态接口,用于输出该先进先出(FIFO)缓存中存储的自适应多速率(AMR) 编码帧队列的长度和编码帧的类型,  a storage status interface for outputting the length of the adaptive multi-rate (AMR) encoded frame queue stored in the first in first out (FIFO) buffer and the type of the encoded frame,
一控制单元, 用于根据删除指令删除存储在先进先出 (FIFO)缓存内的自适应多速 率(AMR)编码帧队列中的无数据(NO— DATA)类型的自适应多速率(AMR)编码帧。 为了减小由自适应多速率(AMR)帧延迟造成的影响,可以在所述先进先出(FIFO) 缓存存储 2个以上的自适应多速率(AMR)编码帧后开始向自适应多速率(AMR)译码 器输入自适应多速率 (AMR)编码帧。 a control unit, configured to delete the adaptive multi-speed stored in the first in first out (FIFO) buffer according to the delete instruction Rate-of-Availability (AMR) encoded non-data (NO-DATA) type of adaptive multi-rate (AMR) coded frames in a frame queue. In order to reduce the impact caused by adaptive multi-rate (AMR) frame delay, adaptive multi-rate can be started after storing more than 2 adaptive multi-rate (AMR) coded frames in the first in first out (FIFO) buffer ( The AMR) decoder inputs an adaptive multi-rate (AMR) coded frame.
对于发送自适应多速率(AMR)编码帧的无线网络控制器(RNC)来说, 它可以使 用接收方移动台的接收缓存区下限值来控制切换时它在发送缓存区内保留的自适应多速 率 (AMR)编码序列的个数, 因此需要一种机制来传递接收方移动台的接收缓存区下限 值, 本发明提出移动台向网络侧发送含所述缓存区下限值的消息的方法, 以此方法向核 心网和无线网络控制器(RNC)传递该下限值, 网络侧延迟方式发送时以此来控制时延, 移动台也可在以间断方式发送方案的缓存区的限定值设定时以此为参考。  For a Radio Network Controller (RNC) that transmits an Adaptive Multi-Rate (AMR) coded frame, it can use the receiver's mobile station's receive buffer lower-limit value to control its adaptation in the transmit buffer when switching. The number of multi-rate (AMR) coding sequences, therefore, a mechanism is needed to transmit the receiving buffer lower limit value of the receiving mobile station, and the present invention proposes that the mobile station sends a message containing the buffer lower limit value to the network side. In this way, the lower limit value is transmitted to the core network and the radio network controller (RNC), and the network side delay mode is sent to control the delay, and the mobile station can also limit the buffer area of the scheme in the intermittent manner. This is used as a reference when setting the value.
——移动台中以间断方式生成和传送自适应多速率(AMR) 编码帧的方法。  - A method of generating and transmitting adaptive multi-rate (AMR) coded frames in a discontinuous manner in a mobile station.
移动台有间断的发送的技术方案是: 按照上述的在移动台中生成和传送自适应多速 率(AMR)编码帧方法, 在移动台暂停发送承载话音数据的无线帧期间对来自自适应多 速率(AMR)编码器的话音帧编码序列进行缓存, 在所述暂停结束后, 在传送话音帧编 码序列中的自适应多速率 (AMR)编码帧到传输信道的过程中, 以每 20毫秒调度一个 以上自适应多速率 (AMR) 编码帧到传输信道的方式, 处理被缓存的部分或全部话音帧 编码序列。  The technical solution for the intermittent transmission of the mobile station is: according to the above-mentioned method for generating and transmitting an adaptive multi-rate (AMR) coded frame in the mobile station, during the time when the mobile station pauses to transmit the radio frame carrying the voice data from the adaptive multi-rate ( The AMR) encoder's voice frame coding sequence is buffered, and after the pause is completed, more than one every 20 milliseconds is scheduled during the transmission of the adaptive multi-rate (AMR) coded frame in the voice frame coding sequence to the transmission channel. Adaptive Multi-Rate (AMR) A method of encoding a frame to a transmission channel, processing some or all of the voice frame coding sequences that are buffered.
现有技术使用的是固定的 20毫秒传输时间间隔 (TTI) 的传输格式组合, 以用该组 合中的一个传输格式调度一个自适应多速率 (AMR)帧的一个类的比特到该格式的传输 信道上的方式发送自适应多速率(AMR)帧,本发明将不再采取固定传输时间间隔(TTI) 的方式, 取而代之的是, 为所述的需要发送的待处理的话音帧编码序列选择传输时间间 隔(ΤΉ)可变的传输格式组合, 例如用传输时间间隔(TTI)为 10毫秒的传输格式组合 去传输一个话音帧编码序列中的一个模式的自适应多速率(AMR) 帧。  The prior art uses a fixed 20 millisecond transmission time interval (TTI) transport format combination to schedule a class of adaptive multi-rate (AMR) frames to a transmission of the format using one of the combinations. The method on the channel transmits an adaptive multi-rate (AMR) frame, and the present invention will no longer adopt a fixed transmission time interval (TTI), but instead selects a transmission sequence of the to-be-processed voice frame to be transmitted. A time interval (ΤΉ) variable transport format combination, such as a transport format combination with a transmission time interval (TTI) of 10 milliseconds to transmit an adaptive multi-rate (AMR) frame of one of the voice frame coding sequences.
所以, 在前面所述的带先进先出 (FIFO) 的移动台中, 可以根据先进先出 (FIFO) 中存储的话音帧编码序列的数量来决定传输时间间隔(TTI)的值, 一种方法是当所述先 进先出 (FIFO) 中存储的话音帧编码序列的数量超过指定的数目时, 优选传输时间间隔 (TTI) 为 10毫秒的传输格式组合, 本发明建议所述指定的数目为 1。  Therefore, in the above-mentioned mobile station with first in first out (FIFO), the value of the transmission time interval (TTI) can be determined according to the number of voice frame coding sequences stored in the first in first out (FIFO), one method is When the number of voice frame coding sequences stored in the first in first out (FIFO) exceeds a specified number, a transmission format combination having a transmission time interval (TTI) of 10 milliseconds is preferred, and the present invention suggests that the specified number is one.
移动台暂停发送无线帧的情形有多种, 都属于暂停发送承载话音数据的无线帧的范 畴, 小区切换是最常见的, 其它的还有: 动态信道调整, 即移动台从一个物理信道切换 到另一物理信道。 偷帧期间移动台虽然不暂停发送无线帧, 但它属于暂停发送承载话音 数据的无线帧的范畴。  There are many situations in which a mobile station pauses to transmit a radio frame, which belongs to the category of a radio frame that suspends transmission of voice data. Cell handover is the most common, and other: dynamic channel adjustment, that is, the mobile station switches from one physical channel to Another physical channel. Although the mobile station does not suspend the transmission of the radio frame during the stealing of the frame, it belongs to the category of the radio frame in which the transmission of the voice data is suspended.
所述移动台暂停发送无线帧期间指小区切换期间时, 即, 在小区切换期间对来自自 适应多速率(AMR) 编码器的话音帧编码序列进行缓存, 在切换完成后, 在传送话音帧 编码序列中的自适应多速率(AMR) 编码帧到传输信道的过程中, 以每 20毫秒调度一 个以上 (不包括一个) 自适应多速率 (AMR) 编码帧到传输信道的方式, 处理被缓存的 部分或全部话音帧编码序列。 When the mobile station pauses to transmit the radio frame period during the cell handover period, that is, buffers the voice frame coding sequence from the adaptive multi-rate (AMR) encoder during the cell handover, after the handover is completed, the voice frame coding is performed. Adaptive Multi-Rate (AMR) in the sequence to encode a frame to the transmission channel, scheduling one every 20 milliseconds More than one (excluding one) adaptive multi-rate (AMR) coded frame to transport channel, processing part or all of the voice frame coding sequence being buffered.
缓存小区切换期间无法发送的自适应多速率 (AMR)编码帧, 在切换成功后, 将这 些缓存的自适应多速率 (AMR)编码帧以每 20毫秒调度一个以上的速率发送, 就能实 现切换期间无暂停话音放送的现象, 这就是这种方法所带来的好处是, 它克服了切换引 起的话音帧丢失所带给听者的不适效应。  An adaptive multi-rate (AMR) coded frame that cannot be transmitted during a cell handover. After the handover is successful, the buffered adaptive multi-rate (AMR) coded frame is transmitted at a rate of more than one every 20 milliseconds. There is no pause in voice delivery during this period. This is the advantage of this method, which overcomes the discomfort that the voice frame loss caused by the handover brings to the listener.
当所述移动台暂停发送无线帧期间指在优先调度高于话音逻辑信道优先级的其它逻 辑信道到传送信道时的暂停话音数据发送期间时, 就意味着, 移动台在被高优先级逻辑 信道中断话音数据发送时的有缓存的发送的技术方案是: 按照上述的在移动台中生成和 传送自适应多速率 (AMR)编码帧方法, 在优先调度高于话音逻辑信道优先级的其它逻 辑信道到传送信道时的暂停话音数据发送期间, 对来自自适应多速率 (AMR)编码器的 话音帧编码序列进行缓存, 话音数据发送恢复后, 在传送话音帧编码序列中的自适应多 速率(AMR) 编码帧到传输信道的过程中, 以每 20毫秒调度一个以上(不包括一个) 自适应多速率 (AMR) 编码帧到传输信道的方式, 处理被缓存的部分或全部话音帧编码 序列。  When the mobile station pauses to transmit a radio frame during a paused voice data transmission period when preferentially scheduling other logical channels higher than the voice logical channel priority to the transport channel, it means that the mobile station is in the high priority logical channel. The technical solution for interrupting the transmission of the buffered voice data is: according to the above method of generating and transmitting an adaptive multi-rate (AMR) coded frame in the mobile station, prioritizing the scheduling of other logical channels higher than the priority of the voice logical channel to During the pause of voice data transmission on the transport channel, the voice frame coding sequence from the adaptive multi-rate (AMR) encoder is buffered, and the adaptive multi-rate (AMR) in the transmitted voice frame coding sequence is recovered after the voice data transmission is restored. In the process of encoding a frame to a transmission channel, one or more (excluding one) adaptive multi-rate (AMR) coded frames are transmitted to the transmission channel every 20 milliseconds, and part or all of the voice frame coding sequences that are buffered are processed.
缓存被调度高优先级逻辑信道所打断发送的话音逻辑信道上的自适应多速率 (AMR)编码帧, 在物理信道速率提高或者高优先级逻辑信道停止发送后, 将这些缓存 的自适应多速率 (AMR) 编码帧以每 20毫秒调度一个以上的速率发送, 就能解决这类 由逻辑信道优先级所带来的话音数据被偷帧的问题, 这就是这种方法所带来的好处。  The buffer is interrupted by the scheduled high-priority logical channel to intercept the adaptive multi-rate (AMR) encoded frame on the voice logical channel. After the physical channel rate is increased or the high-priority logical channel stops transmitting, the cache is adaptive. Rate (AMR) coded frames are transmitted at more than one rate every 20 milliseconds, which solves the problem of such framed data being stolen by the logical channel priority. This is the benefit of this method.
上述移动台有间断的发送的技术方案中, 以及移动台在切换时有缓存的发送的技术 方案和移动台在被高优先级逻辑信道中断话音数据发送时的有缓存的发送的技术方案 中, 都有每 20毫秒调度一个以上 (不包括一个) 自适应多速率 (AMR)编码帧到传输 信道的方式, 实现这一方式的方法有多种, 本发明给出一个具体的方法, 即, 所述每 20 毫秒调度一个以上 (不包括一个) 自适应多速率 (AMR)编码帧到传输信道的方式是, 调度一个完整的自适应多速率(AMR)编码帧到 10毫秒传输时间间隔 (TTI) 的传输信 道。 将一个完整的自适应多速率(AMR) 编码帧调度到 10毫秒传输时间间隔 (ΤΉ) 的 传输信道上的方法, 采取 10毫秒的传输格式(TF), 将一个自适应多速率(AMR)编码 帧的所有类均调度到传输信道上。  In the technical solution that the mobile station has intermittent transmission, and the technical solution that the mobile station has buffered transmission when switching, and the technical solution that the mobile station transmits the buffer when the voice data is interrupted by the high priority logical channel, There are multiple ways to schedule more than one (not including one) adaptive multi-rate (AMR) coded frame to the transmission channel every 20 milliseconds. There are various ways to implement this method. The present invention provides a specific method, that is, One or more (excluding one) adaptive multi-rate (AMR) coded frames are transmitted to the transmission channel every 20 milliseconds by scheduling a complete adaptive multi-rate (AMR) coded frame to a 10 millisecond transmission time interval (TTI). Transport channel. A method of scheduling a complete adaptive multi-rate (AMR) coded frame onto a 10 millisecond transmission time interval (ΤΉ) transmission channel, taking a 10 millisecond transmission format (TF), and an adaptive multi-rate (AMR) coding All classes of frames are scheduled onto the transport channel.
网络侧的无线网络控制器(RNC)检测接收到的自适应多速率 (AMR)编码帧是否 是被延迟的帧, 一种检测的方法就是检査每 20毫秒收到的自适应多速率(AMR)编码 帧, 一旦发现 20毫秒内收到 1个以上的帧或者 10毫秒内就收到一个帧, 就表明收到了 被延迟的帧; 另一种检测的方法就是向无线接入网发送信令消息, 该消息指明若干个被 延迟帧的发送无线帧的起始帧号和帧个数。 无线网络控制器 (RNC) 向核心网发送带延 迟标志的自适应多速率 (AMR)编码帧的一种方法是: 将延迟标志可以放在承载该自适 应多速率(AMR) 编码帧的核心网-无线接入网接口 (lu) 用户平面帧的扩展域 (spare extension)里。 The radio network controller (RNC) on the network side detects whether the received adaptive multi-rate (AMR) coded frame is a delayed frame. One method of detection is to check the adaptive multi-rate (AMR) received every 20 milliseconds. The coded frame, once it is found that more than one frame is received within 20 milliseconds or a frame is received within 10 milliseconds, indicating that the delayed frame is received; another method of detecting is to send signaling to the radio access network. A message indicating the starting frame number and the number of frames of the transmitted radio frame of a number of delayed frames. A method for a Radio Network Controller (RNC) to transmit an adaptive multi-rate (AMR) coded frame with a delay flag to the core network is: placing a delay flag on the bearer Should be multi-rate (AMR) encoded frame core network - radio access network interface (lu) user plane frame extension field (spare extension).
——网络侧以间断的方式传送自适应多速率 (AMR) 编码帧的方法。  - The method of transmitting adaptive multi-rate (AMR) coded frames in an intermittent manner on the network side.
通用移动通信*** (UMTS ) 网络侧的有间断的发送的技术方案是: 按照上述的在 通用移动通信*** (UMTS) 的网络侧的一种生成和传送自适应多速率 (AMR) 编码帧 的方法, 以及在通用移动通信*** (UMTS) 网络侧由无线网络控制器 (RNC)确定编 码命令中模式的编码生成和发送的方法, 无线网络控制器 (RNC) 设立缓存区, 将暂停 无线帧发送期间收到的由核心网发送的 lu用户平面帧所承载的话音帧编码序列缓存在该 缓存区, 所述暂停结束后, 在调度被缓存的话音帧编码序列中的自适应多速率(AMR) 编码帧到传输信道的过程中, 采用每 20毫秒调度一个以上(不包括一个) 自适应多速率 (AMR)编码帧到传输信道的方式。  A discontinuous transmission scheme on the network side of the Universal Mobile Telecommunications System (UMTS) is: a method of generating and transmitting an adaptive multi-rate (AMR) coded frame on the network side of the Universal Mobile Telecommunications System (UMTS) as described above. And a method of determining, by the radio network controller (RNC), a code generation and transmission of a mode in the coding command on a universal mobile communication system (UMTS) network side, the radio network controller (RNC) setting a buffer area, and suspending the radio frame transmission period The received voice frame coding sequence carried by the lu user plane frame sent by the core network is buffered in the buffer area, and after the pause is finished, the adaptive multi-rate (AMR) coding in the buffered voice frame coding sequence is scheduled. In the process of frame-to-transport channel, one or more (excluding one) adaptive multi-rate (AMR) coded frames are transmitted to the transmission channel every 20 milliseconds.
在小区切换时的通用移动通信*** (UMTS ) 网络侧发送的技术方案是: 按照通用 移动通信*** (UMTS ) 网络侧的有间断的发送方法, 所述暂停无线帧发送期间是由无 线网络控制器 (RNC)之内的小区间切换期间。  The technical solution for transmitting on the network side of the Universal Mobile Telecommunications System (UMTS) at the time of cell handover is: according to the discontinuous transmission method on the network side of the Universal Mobile Telecommunications System (UMTS), the paused radio frame transmission period is determined by the radio network controller Inter-cell handover period within (RNC).
在无线网络控制器 (RNC)间切换时的通用移动通信***(UMTS)网络侧发送的技术 方案是: 按照通用移动通信*** (UMTS ) 网络侧的有间断的发送方法, 所述暂停无线 帧发送期间是无线网络控制器 (RNC)间切换期间。  The technical solution of the Universal Mobile Telecommunications System (UMTS) network side transmission when switching between Radio Network Controllers (RNCs) is: According to the discontinuous transmission method on the network side of the Universal Mobile Telecommunications System (UMTS), the suspended radio frame transmission The period is during the switching between radio network controllers (RNCs).
在被高优先级逻辑信道中断话音数据发送时的通用移动通信*** (UMTS) 网络侧 的发送的技术方案是: 按照通用移动通信*** (UMTS) 网络侧的有间断的发送方法, 所述暂停无线帧发送期间是由优先调度高于话音逻辑信道优先级的其它逻辑信道到传送 信道时的暂停话音数据发送所引起的未能发送那些核心网-无线接入网接口 (lu)用户平 面帧所承载的话音帧编码序列中自适应多速率 (AMR)编码帧的期间。  The technical solution for transmitting on the universal mobile communication system (UMTS) network side when the voice data is interrupted by the high priority logical channel is: according to the intermittent transmission method on the network side of the Universal Mobile Telecommunications System (UMTS), the pause wireless The frame transmission period is caused by the failure to transmit those core network-radio access network interface (lu) user plane frames caused by the suspension of voice data transmission when the other logical channels are preferentially scheduled to be higher than the voice logical channel priority to the transmission channel. The period of the adaptive multi-rate (AMR) encoded frame in the voice frame coding sequence.
——网络侧传送被延迟的自适应多速率 (AMR)编码帧的方法。  - A method of transmitting delayed adaptive multi-rate (AMR) coded frames on the network side.
当移动台使用了有间断的发送方法时, 会产生被延迟了的自适应多速率 (AMR)编 码帧, 引起发送移动台的无线网络控制器 (RNC)以每 20毫秒超过一个话音帧编码序列的 速率向核心网发送话音数据,核心网也一定会以每 20毫秒超过一个话音帧编码序列的速 率向接收移动台的无线网络控制器 (RNC)发送,处理这类帧的通用移动通信***(UMTS) 网络侧的发送技术方案是: 按照上述的在通用移动通信*** (UMTS ) 的网络侧的一种 生成和传送自适应多速率 (AMR) 编码帧的方法, 以及在通用移动通信***(UMTS) 网络侧由无线网络控制器(RNC)确定编码命令中模式的编码生成和发送的方法, 所述 核心网发送到无线网络控制器(RNC) 的承载话音帧编码序列的 lu用户平面帧包含该话 音帧编码序列的延迟标志, 当延迟标志指示话音帧编码序列是被延迟了的时, 无线网络 控制器(RNC) 以每 20毫秒调度一个以上 (不包括一个) 自适应多速率(AMR) 编码 帧到传输信道的方式来调度该话音帧编码序列。 在上述网络侧以间断方式发送自适应多速率(AMR) 编码帧或在传送传送被延迟的 自适应多速率 (AMR) 编码帧的时候都要使用每 20毫秒调度一个以上(不包括一个) 自适应多速率 (AMR) 编码帧到传输信道的方式, 这里给出一个具体的方法, 即, 所述 每 20毫秒调度一个以上(不包括一个) 自适应多速率 (AMR) 编码帧到传输信道的方 式是, 调度一个完整的自适应多速率 (AMR)编码帧到 10毫秒传输时间间隔 (TTI) 的 传输信道。将一个完整的自适应多速率(AMR)编码帧调度到 10毫秒传输时间间隔(ΤΉ) 的传输信道上的方法, 采取 10毫秒的传输格式(TF), 将一个自适应多速率(AMR)编 码帧的所有类均调度到传输信道上。 When the mobile station uses a discontinuous transmission method, a delayed adaptive multi-rate (AMR) coded frame is generated, causing the radio network controller (RNC) transmitting the mobile station to exceed one voice frame coding sequence every 20 milliseconds. The rate of transmission of voice data to the core network, the core network will also transmit to the radio network controller (RNC) of the receiving mobile station at a rate of more than one voice frame coding sequence every 20 milliseconds, a universal mobile communication system that processes such frames ( The transmission scheme of the UMTS) network side is: a method for generating and transmitting an adaptive multi-rate (AMR) coded frame on the network side of the Universal Mobile Telecommunications System (UMTS) as described above, and in a universal mobile communication system (UMTS) a method for determining, by a radio network controller (RNC), a code generation and transmission of a mode in an encoding command, the lu user plane frame transmitted by the core network to a radio network controller (RNC) carrying a voice frame coding sequence, The delay flag of the voice frame coding sequence, when the delay flag indicates that the voice frame coding sequence is delayed, the radio network controller (R NC) The speech frame coding sequence is scheduled by scheduling more than one (excluding one) adaptive multi-rate (AMR) coded frame to the transmission channel every 20 milliseconds. To transmit an adaptive multi-rate (AMR) coded frame in a discontinuous manner on the above network side or to schedule more than one (excluding one) every 20 milliseconds when transmitting an adaptively multi-rate (AMR) coded frame that is delayed in transmission. Adapting to the multi-rate (AMR) coding frame to the transmission channel, a specific method is given here, that is, scheduling more than one (excluding one) adaptive multi-rate (AMR) coded frame to the transmission channel every 20 milliseconds. The method is to schedule a complete adaptive multi-rate (AMR) coded frame to a 10 millisecond transmission time interval (TTI) transmission channel. A method of scheduling a complete adaptive multi-rate (AMR) coded frame onto a 10 millisecond transmission time interval (ΤΉ) transmission channel, using a 10 millisecond transmission format (TF), and an adaptive multi-rate (AMR) coding All classes of frames are scheduled onto the transport channel.
在上述的通用移动通信*** (UMTS) 网络侧的有间断的发送的技术方案中, 以及 在相关的小区切换时、无线网络控制器 (RNC)间切换时,被高优先级逻辑信道中断话音数 据发送时、和核心网以每 20毫秒超过一个话音帧编码序列的速率向无线网络控制器 (RNC) 发送话音数据时, 需要对话音的延时进行控制, 因此对缓存在无线网络控制器 (RNC) 缓存区内的话音帧编码序列的数目进行限制, 具体的方法是, 当缓存的话音帧编码序列 的数目超过目的移动台的缓存区下限值后,开始丢弃缓存的话音帧编码序列。  In the above-mentioned technical solution of intermittent transmission on the network side of the Universal Mobile Telecommunications System (UMTS), and when switching between radio network controllers (RNCs) during the relevant cell handover, the voice data is interrupted by the high priority logical channel. When transmitting, and when the core network sends voice data to the Radio Network Controller (RNC) at a rate of more than one voice frame coding sequence every 20 milliseconds, the delay of the voice is required to be controlled, so the buffer is stored in the radio network controller (RNC). The number of voice frame coding sequences in the buffer area is limited. When the number of buffered voice frame coding sequences exceeds the lower limit of the buffer area of the destination mobile station, the buffered voice frame coding sequence is discarded.
在上述所有的技术方案中, 无论是移动台还是网络侧使用上述的有间断的发送方式 都需要将每个 20毫秒的话音帧编码为多个自适应多速率 (AMR)模式的帧, 因为经过 缓存延迟后, 它所要面对的信道可能有变化。 所以说多自适应多速率 (AMR)模式是间 断发送所需要的技术特征。  In all the above technical solutions, whether the mobile station or the network side uses the above-mentioned intermittent transmission mode, it is required to encode each 20-millisecond voice frame into multiple adaptive multi-rate (AMR) mode frames, because After the buffer is delayed, the channel it faces may change. So multi-adaptive multi-rate (AMR) mode is a technical feature required for intermittent transmission.
本发明使得在 TFC选择阶段有了自适应多速率(AMR)模式的选择机制, 减少现有 技术中在 TFC选择阶段因单一自适应多速率(AMR)模式同 TF不匹配引起的话音帧丢 失, 本发明中, 自适应多速率 (AMR) 编码器的输出是多模式的, 并且这种多模式有传 输信道格式组合与之相匹配。 对于因物理信道速率突然下降或者调度突发高优先级逻辑 信道引起的有效传输信道格式组合 (TFC) 突然变化, 本发明的方法中多模式的自适应 多速率(AMR) 编码器的输出可以用来匹配所述的有效传输信道格式组合(TFC)突然 变化, 从而减少因物理信道速率突然下降或者调度突发高优先级逻辑信道引起的话音帧 的丢帧。  The present invention enables an adaptive multi-rate (AMR) mode selection mechanism in the TFC selection phase, which reduces the loss of voice frames caused by the single adaptive multi-rate (AMR) mode and the TF mismatch in the TFC selection phase in the prior art. In the present invention, the output of the adaptive multi-rate (AMR) encoder is multi-mode, and this multi-mode has a combination of transport channel formats to match. The output of the multi-mode adaptive multi-rate (AMR) encoder in the method of the present invention can be used for abrupt changes in the effective transport channel format combination (TFC) caused by a sudden drop in the physical channel rate or scheduling of burst high priority logical channels. To match the described effective transmission channel format combination (TFC) abrupt changes, thereby reducing frame loss of voice frames due to a sudden drop in physical channel rate or scheduling of burst high priority logical channels.
本发明中, 自适应多速率(AMR)编码器输出在先进先出 (FIFO)缓存里的是多模 式的自适应多速率 (AMR)编码帧, 并且这种多模式有传输信道格式组合与之相匹配。 对于因物理信道速率突然下降、 切换或者调度突发高优先级逻辑信道引起的有效传输信 道格式组合(TFC) 的突然变化, 即传输信道格式组合集合(TFCS) 中可以使用的传输 信道格式组合不再包含这之前移动台所使用的传输信道格式组合(TFC)。本发明的方法 中多模式的自适应多速率(AMR)编码器的输出可以用来匹配所述的有效传输信道格式 组合 (TFC) 的突然变化, 即用新的可用的传输信道格式组合 (TFC) 来调度先进先出 (FIFO)缓存里待处理的话音帧编码序列中与之前移动台发送的自适应多速率 (AMR) 模式不同的那种模式的自适应多速率 (AMR) 编码帧, 从而减少因物理信道速率突然下 降、 切换或者调度突发高优先级逻辑信道引起的话音帧的丢帧, 与现有技术相比本发明 能在更短的时间内针对所有通用移动通信***(UMTS) 的自适应多速率 (AMR)模式 和移动台的无线资源选择最佳传输信道格式组合(TFC)。 In the present invention, the adaptive multi-rate (AMR) encoder outputs a multi-mode adaptive multi-rate (AMR) coded frame in a first-in, first-out (FIFO) buffer, and the multi-mode has a transport channel format combination thereof. Match. Abrupt changes in the effective transport channel format combination (TFC) caused by a sudden drop in the physical channel rate, handover, or scheduling of bursty high priority logical channels, ie, the transport channel format combination that can be used in the Transport Channel Format Combination Set (TFCS) This includes the Transport Channel Format Combination (TFC) used by the mobile station before this. The output of the multi-mode adaptive multi-rate (AMR) encoder in the method of the present invention can be used to match the abrupt changes in the effective transport channel format combination (TFC), ie, using a new available transport channel format combination (TFC) ) to schedule the adaptive multi-rate (AMR) transmitted by the previous mobile station in the sequence of voice frames to be processed in the first-in, first-out (FIFO) buffer Adaptive multi-rate (AMR) coded frames of different modes of mode, thereby reducing frame loss of voice frames caused by sudden drops in physical channel rates, handover or scheduling of bursty high priority logical channels, compared to the prior art The present invention enables the selection of the best transport channel format combination (TFC) for the adaptive multi-rate (AMR) mode of all Universal Mobile Telecommunications System (UMTS) and the radio resources of the mobile station in a shorter time.
本发明中的缓存机制克服了现有技术中的每 20 毫秒必须完成一自适应多速率 (AMR)话音帧发送或丢弃这一限制, 因为使用本发明的方法可以做到延迟发送, 这样 使得切换和调度突发高优先级逻辑信道抢占期间的自适应多速率 (AMR)编码器输出的 编码帧仍能得到传送, 而现有技术在会丢弃这些被切换和调度突发高优先级逻辑信道抢 占了无线资源的自适应多速率 (AMR) 编码帧, 即在现有技术中在切换和调度突发高优 先级逻辑信道抢占无线资源的期间会发生在存储单元中的自适应多速率(AMR) 编码帧 未被读取就被刷新的现象。  The buffering mechanism of the present invention overcomes the limitation that an adaptive multi-rate (AMR) voice frame transmission or discarding must be completed every 20 milliseconds in the prior art, because delayed transmission can be achieved using the method of the present invention, thus making handover The coded frames output by the adaptive multi-rate (AMR) encoder during the scheduling of the burst high-priority logical channel preemption can still be transmitted, and the prior art will discard these switched and scheduled burst high-priority logical channel preemptions. An adaptive multi-rate (AMR) coded frame of a radio resource, that is, an adaptive multi-rate (AMR) that occurs in a storage unit during handover and scheduling of burst high-priority logical channels to preempt wireless resources in the prior art. The phenomenon that the encoded frame is refreshed without being read.
与现有技术中偷帧不同, 本发明方法中那些受调度突发高优先级逻辑信道影响的话 音帧可不受延迟地用与它前面相邻的话音帧不同的模式发送,或者延迟一定时间后发送, 不会引起话音帧的丢弃。 这体现在它可以调整先进先出 (FIFO) 缓存读取自适应多速率 (AMR)编码帧的定时, 在先进先出 (FIFO)缓存写入和读取时间间隔大于 10毫秒时, 可以用 10毫秒传输时间间隔 (TTI) 的传输格式组合发送在先进先出 (FIFO) 缓存中的 自适应多速率 (AMR)编码帧以减小所述的写入和读取的时间间隔。  Unlike the prior art stealing frames, the voice frames affected by the scheduled burst high priority logical channels in the method of the present invention may be transmitted in a different mode than the voice frames adjacent thereto before being delayed, or delayed after a certain time. Sending does not cause discarding of voice frames. This is reflected in the fact that it can adjust the timing of the first-in, first-out (FIFO) cache read adaptive multi-rate (AMR) coded frame. When the first-in-first-out (FIFO) buffer write and read time interval is greater than 10 ms, 10 can be used. The transmission format of the millisecond transmission time interval (TTI) combines the adaptive multi-rate (AMR) encoded frames transmitted in the first in first out (FIFO) buffer to reduce the time interval between the writing and reading.
本发明给出的自适应多速率 (AMR)编码的生成和传送的方法在最大程度上利用了 物理信道所提供的比特速率来传送话音数据,这种利用体现在对 20毫秒的话音输入可进 行在一定模式范围内和一定的时间范围内选择, 和这种选择对物理信道的适应上。 附图说明  The method for generating and transmitting adaptive multi-rate (AMR) coding according to the present invention maximizes the use of the bit rate provided by the physical channel to transmit voice data. This utilization is reflected in the 20 millisecond voice input. Select within a certain mode range and within a certain time range, and the adaptation of this choice to the physical channel. DRAWINGS
图 1是移动台生成和传送自适应多速率 (AMR) 编码帧的实施例的方框图。  1 is a block diagram of an embodiment of a mobile station generating and transmitting an adaptive multi-rate (AMR) coded frame.
图 2是图 1中编码命令中的自适应多速率(AMR)模式数为 2的编码模块的处理示 意图。 .  Figure 2 is a process diagram of an encoding module having an adaptive multi-rate (AMR) mode number of 2 in the encoding command of Figure 1. .
图 3是图 1中编码命令中的自适应多速率(AMR)模式数为 3的编码模块的处理示 意图。  Figure 3 is a process diagram of an encoding module having an adaptive multi-rate (AMR) mode number of 3 in the encoding command of Figure 1.
图 4是移动台以缓存方式生成和传送自适应多速率 (AMR) 编码帧的实施例的方框 图。  4 is a block diagram of an embodiment of a mobile station generating and transmitting adaptive multi-rate (AMR) coded frames in a buffered manner.
图 5是自适应多速率(AMR) 编码器的先进先出 (FIFO) 缓存的接口方框图。  Figure 5 is an interface block diagram of a first-in, first-out (FIFO) buffer for an adaptive multi-rate (AMR) encoder.
图 6〜图 10是移动台在表格 1 中所示的 0到 80毫秒的这段时间内处理先进先出 (FIFO)缓存中的话音帧编码序列的示意图; 其中  Figure 6 to Figure 10 are diagrams showing the processing of the voice frame coding sequence in the first in first out (FIFO) buffer during the period from 0 to 80 milliseconds shown in Table 1 by the mobile station;
图 6是在 0毫秒先进先出 (FIFO)缓存输出自适应多速率(AMR)编码帧前的示意 图; 图 7是在 0毫秒先进先出 (FIFO)缓存输出自适应多速率(AMR)编码帧时的示意 图; 6 is a schematic diagram of a zero millisecond first in first out (FIFO) buffer output adaptive multi-rate (AMR) coded frame; 7 is a schematic diagram of outputting an adaptive multi-rate (AMR) encoded frame in a 0 millisecond first in first out (FIFO) buffer;
图 8表示 35毫秒时自适应多速率(AMR)编码器向先进先出 (FIFO) 缓存输出话 音帧编码序列的示意图;  Figure 8 is a diagram showing an adaptive multi-rate (AMR) encoder outputting a sequence of voice frame codes to a first in first out (FIFO) buffer at 35 milliseconds;
图 9是在 40毫秒先进先出 (FIFO)缓存输出自适应多速率(AMR)编码帧前的示 意图;  Figure 9 is an illustration of a 40 millisecond first-in, first-out (FIFO) buffer output adaptive multi-rate (AMR) coded frame;
图 10是在 40毫秒先进先出 (FIFO)缓存输出自适应多速率(AMR)编码帧时的示 意图。  Figure 10 is a schematic illustration of a 40 millisecond first-in, first-out (FIFO) buffer output adaptive multi-rate (AMR) coded frame.
图 11是移动台的延迟译码的实施例的方框图。  11 is a block diagram of an embodiment of delay decoding by a mobile station.
图 12是自适应多速率 (AMR) 译码器的先进先出 (FIFO) 缓存的接口方框图。 图 13〜图 16是移动台处理连续 5个自适应多速率 (AMR)编码帧的示意图; 其中 图 13是发送无数据 (NO— DATA)帧到先进先出 (FIFO) 缓存时的示意图; 图 14是收到无数据 (NO_DATA)帧后先进先出(FIFO)缓存发送自适应多速率 (AMR) 帧到变速率译码器的示意图;  Figure 12 is an interface block diagram of a first-in, first-out (FIFO) buffer of an adaptive multi-rate (AMR) decoder. 13 to FIG. 16 are schematic diagrams of a mobile station processing five consecutive adaptive multi-rate (AMR) coded frames; FIG. 13 is a schematic diagram of transmitting a no-data (NO-DATA) frame to a first-in-first-out (FIFO) buffer; 14 is a schematic diagram of a first-in-first-out (FIFO) buffer transmission adaptive multi-rate (AMR) frame to variable rate decoder after receiving a no-data (NO_DATA) frame;
图 15是先进先出(FIFO)缓存接收到 2个 10毫秒传输时间间隔(ΤΉ)的自适应多 速率 (AMR) 帧之后的示意图;  Figure 15 is a schematic diagram of the first in first out (FIFO) buffer after receiving two 10 millisecond transmission time intervals (ΤΉ) of adaptive multi-rate (AMR) frames;
图 16是先进先出(FIFO)缓存接收到 2个 10毫秒传输时间间隔(TTI)的自适应多 速率 (AMR) 帧之后删除一个无数据 (NO— DATA) 帧的示意图。  Figure 16 is a diagram of a first-in, first-out (FIFO) buffer that deletes a no-data (NO-DATA) frame after receiving two 10 ms transmission time interval (TTI) adaptive multi-rate (AMR) frames.
图 17是网络侧生成和传送自适应多速率(AMR)编码帧的实施例的方框图。  17 is a block diagram of an embodiment of a network side generating and transmitting an adaptive multi-rate (AMR) coded frame.
图 18是网络侧以间断的方式传送自适应多速率 (AMR) 编码帧的实施例和网络侧 传送被延迟的自适应多速率 (AMR) 编码帧的实施例的方框图。  Figure 18 is a block diagram of an embodiment of a network side transmitting an adaptive multi-rate (AMR) coded frame in an intermittent manner and a network side transmitting delayed adaptive multi-rate (AMR) coded frame.
图 19是一个通用移动通信*** (UMTS)手机基本功能框图。 具体实施方式  Figure 19 is a basic functional block diagram of a Universal Mobile Telecommunications System (UMTS) handset. detailed description
下面结合附图详细说明本发明的优选实施例。  Preferred embodiments of the present invention will be described in detail below with reference to the accompanying drawings.
实施例 1——移动台生成和传送自适应多速率 (AMR) 编码帧  Embodiment 1 - Mobile Station Generation and Transmission of Adaptive Multi-Rate (AMR) Coded Frames
参见图 1, 如其所示, 在移动台中, 自适应多速率 (AMR)编码器将话音从声音转 换成电信号, 经过滤波后, 再经过采样实现了从模拟话音信号到数字话音信号 1的转换, 编码模块按 20毫秒一帧的方式为数字话音信号 1编码, 编码模块每 20毫秒产生 1个话 音帧编码序列 2, 每个话音帧编码序列包含: 模式指示, 若干个自适应多速率(AMR) 话音编码帧或 1个自适应多速率 (AMR)静默帧。 模式指示给出话音帧编码序列及其自 适应多速率 (AMR)编码帧的情况, SP, 是否为有效的话音帧编码序列, 话音帧编码序 列中的自适应多速率(AMR) 编码帧的个数和每个帧的编码模式。 无线接口功能单元中 的传输格式组合选择模块以自适应多速率(AMR)话音编码器的话音帧编码序列 2以及 分组业务数据 4和信令数据 5为输入, 选定一传输格式组合, 根据该传输格式组合确定 传送话音的传输格式以及该格式对应的话音数据传输块 6。话音帧编码序列 2是有效的话 音帧编码序列时, 活音数据传输块 6就是一个自适应多速率 (AMR) 编码帧, 这个自适 应多速率(AMR)编码帧是话音帧编码序列 2所包含的所有自适应多速率(AMR)话音 编码帧(非静默帧) 中的 1个, 或者是话音帧编码序列 2所包含的那 1个静默帧。 由传 输格式组合选择模块输出的对应分组业务数据 4和信令数据 5的传输块分别是分组数据 传输块 8和信令数据传输块 7。无线接口功能单元的其它模块将话音数据传输块 6、分组 数据传输块 8和信令数据传输块 7映射到物理信道上进行发送。 无线接口功能单元中有 一负责向自适应多速率 (AMR) 编码器输出模式选择信号 9的话音帧模式控制模块, 模 式选择信号 9输出到编码模块, 当模式选择信号 9包含多个自适应多速率(AMR)编码 模式时, 模式选择信号 9作为编码命令指示编码模块在输出非静默帧时应包含多个自适 应多速率 (AMR)话音编码帧, 并指示了这些帧的模式。 Referring to FIG. 1, as shown therein, in a mobile station, an adaptive multi-rate (AMR) encoder converts voice from sound to an electrical signal, and after filtering, is sampled to convert from an analog voice signal to a digital voice signal 1. The encoding module encodes the digital voice signal 1 in a manner of 20 milliseconds one frame. The encoding module generates one voice frame coding sequence 2 every 20 milliseconds. Each voice frame coding sequence includes: mode indication, several adaptive multi-rates (AMR) A voice coded frame or an adaptive multi-rate (AMR) silence frame. The mode indication gives the case of the voice frame coding sequence and its adaptive multi-rate (AMR) coded frame, SP, whether it is a valid voice frame coding sequence, and the adaptive multi-rate (AMR) coded frame in the voice frame coding sequence. Number and encoding mode for each frame. The transport format combination selection module in the wireless interface functional unit uses the voice frame coding sequence 2 of the adaptive multi-rate (AMR) voice coder and The packet service data 4 and the signaling data 5 are inputs, a transport format combination is selected, and the transport format of the transport voice and the voice data transport block 6 corresponding to the format are determined according to the transport format combination. When the voice frame coding sequence 2 is a valid voice frame coding sequence, the voice data transmission block 6 is an adaptive multi-rate (AMR) coded frame, and the adaptive multi-rate (AMR) coded frame is included in the voice frame coding sequence 2. One of all adaptive multi-rate (AMR) speech encoded frames (non-silent frames), or the one silence frame contained in the speech frame encoding sequence 2. The transport blocks of the corresponding packet service data 4 and the signaling data 5 output by the transport format combination selection module are the packet data transport block 8 and the signaling data transport block 7, respectively. The other modules of the wireless interface function unit map the voice data transmission block 6, the packet data transmission block 8, and the signaling data transmission block 7 onto the physical channel for transmission. The wireless interface function unit has a voice frame mode control module for outputting the mode selection signal 9 to the adaptive multi-rate (AMR) encoder, the mode selection signal 9 is output to the encoding module, and the mode selection signal 9 includes a plurality of adaptive multi-rates. In the (AMR) coding mode, the mode selection signal 9 as an encoding command indicates that the encoding module should include a plurality of adaptive multi-rate (AMR) speech encoded frames when outputting non-silent frames, and indicate the mode of these frames.
表格 1是一个话音帧编码序列 2的组成结构及其比特数分配的示例。  Table 1 is an example of the composition of a voice frame coding sequence 2 and its bit number allocation.
表格 1 Table 1
Figure imgf000019_0001
图 2是可产生 2种自适应多速率 (AMR)模式的帧序列的编码模块.的处理示意图, 数字话音信号 1同时向带话音激活检测的话音编码功能模块 100和语音编码模块 101输 出,带话音激活检测的话音编码功能模块 100同通用移动通信*** (UMTS)标准 TS26.071 (或 TS26.171 ) 的图 1话音处理功能示意(Overview of audio processing fimctions) 中的 对应部分的结构基本相同, 区别是图 2中画出了输出到带话音激活检测的话音编码功能 模块 100的语音编码子模块的编码模式指示信号 17。 话音激活检测模块输出的是话音激 活检测标志 (Voice Activity Detector flag) 10, 带话音激活检测的话音编码功能模块 100 的语音编码模块输出话音编码帧 12,语音编码模块 101输出的是话音编码帧 19,话音编 码帧 12和 19的每帧比特数分别取决于编码模式指示信号 17和 18, 编码模式指示信号 17和 18是编码模式控制模块将模式选择信号 9中的 2个模式分解成的 1个模式 1路的 2 路输出。不连续发送和操作模块输出自适应多速率(AMR)帧类型信号 11到多路语音编 码复用模块和语音编码模块, 自适应多速率 (AMR) 帧类型信号 11指示: 信息比特 14 是否是有效的自适应多速率编码帧, 自适应多速率编码帧的模式, 所述自适应多速率编 码帧的模式为非静默话音帧的模式或静默帧的模式, 当自适应多速率编码帧的模式是静 默帧时,信息比特 14是舒适噪声发送模块输出的静默检测帧 13。语音编码模块 101输出 自适应多速率话音编码帧 19及其自适应多速率 (AMR)帧类型信号 16, 自适应多速率 (AMR) 帧类型信号 16表示自适应多速率话音编码帧 19的模式。 多路语音编码复用模 块将来自不连续发送和操作模块的信息比特 14和自适应多速率话音编码帧 19合并成话 音帧编码序列 2, 合并的方法是: 当自适应多速率(AMR)帧类型信号 11指示信息比特 14无效时, 设置话音帧编码序列 2的模式指示为无效的话音帧编码序列; 当自适应多速 率(AMR)帧类型信号 11指示信息比特 14是静默帧时, 设置话音帧编码序列 2的模式 指示为静默帧, 将信息比特 14放到话音帧编码序列 2的话音帧编码序列里; 当自适应多 速率 (AMR)帧类型信号 11指示信息比特 14是非静默帧时, 把信息比特 14和 19一同 放到 i舌音帧编码序列 2的话音帧编码序列里, 同时设置话音帧编码序列 2的模式指示为 自适应多速率(AMR) 帧类型信号 11和 16所指示的模式。
Figure imgf000019_0001
2 is a process diagram of an encoding module that can generate a frame sequence of two adaptive multi-rate (AMR) modes. The digital voice signal 1 is simultaneously output to the voice encoding function module 100 and the voice encoding module 101 with voice activation detection. The voice encoding function module 100 of the voice activation detection has substantially the same structure as the corresponding portion of the Overview of audio processing fimctions of the Universal Mobile Telecommunications System (UMTS) standard TS26.071 (or TS 26.171). The difference is that the coding mode indication signal 17 output to the speech coding sub-module of the speech coding function module 100 with voice activation detection is shown in FIG. The voice activation detection module outputs the voice A voice activity detection Detector flag 10, a voice coding module of the voice coding function module 100 with voice activation detection outputs a voice coding frame 12, and the voice coding module 101 outputs a voice coding frame 19, and voice coding frames 12 and 19 The number of bits per frame depends on the coding mode indication signals 17 and 18, respectively, and the coding mode control signals 17 and 18 are two outputs of one mode one way in which the coding mode control module decomposes the two modes of the mode selection signal 9. The discontinuous transmission and operation module outputs an adaptive multi-rate (AMR) frame type signal 11 to a multi-channel speech coding multiplexing module and a speech coding module, and an adaptive multi-rate (AMR) frame type signal 11 indicates: whether the information bit 14 is valid Adaptive multi-rate encoded frame, mode of adaptive multi-rate encoded frame, mode of the adaptive multi-rate encoded frame is a mode of a non-silent speech frame or a mode of a silent frame, when the mode of the adaptive multi-rate encoded frame is When the frame is silenced, the information bit 14 is the silence detection frame 13 output by the comfort noise transmitting module. The speech encoding module 101 outputs an adaptive multi-rate speech encoded frame 19 and its adaptive multi-rate (AMR) frame type signal 16, which represents the mode of the adaptive multi-rate speech encoded frame 19. The multiplexed speech coding multiplexing module combines the information bits 14 from the discontinuous transmission and operation module and the adaptive multi-rate speech coding frame 19 into a speech frame coding sequence 2, the method of combining is: when adaptive multi-rate (AMR) frames When the type signal 11 indicates that the information bit 14 is invalid, the mode of the voice frame coding sequence 2 is set to be an invalid voice frame coding sequence; when the adaptive multi-rate (AMR) frame type signal 11 indicates that the information bit 14 is a silent frame, the voice is set. The mode of the frame coding sequence 2 is indicated as a silence frame, and the information bit 14 is placed in the voice frame coding sequence of the voice frame coding sequence 2; when the adaptive multi-rate (AMR) frame type signal 11 indicates that the information bit 14 is a non-silent frame, The information bits 14 and 19 are placed together in the voice frame coding sequence of the i-tonal frame coding sequence 2, while the mode indication of the voice frame coding sequence 2 is set to be indicated by the adaptive multi-rate (AMR) frame type signals 11 and 16. mode.
表格 2表示一个编码命令中的自适应多速率 (AMR)模式数为 2时编码模块产生的 信息比特 14和自适应多速率话音编码非静默帧 19的构成,分组业务数据 4和信令数据 5 的构成, 以及传输这些用户数据的对应传输信道的标识, 以及相关的逻辑信道的模式和 优先级 (优先级数值越小优先级越髙)。  Table 2 shows the composition of information bits 14 and adaptive multi-rate speech encoded non-silent frames 19 generated by the encoding module when the number of adaptive multi-rate (AMR) modes in an encoding command is 2, packet service data 4 and signaling data 5 The composition, and the identity of the corresponding transport channel for transmitting these user data, and the mode and priority of the associated logical channel (the lower the priority value, the lower the priority).
表格 2 自适应多速肆 ί话音  Table 2 Adaptive multi-speed 肆 ί voice
用户数据 信息比特 14 5 4 编码非静默 ·! 1贞 19  User Data Information Bits 14 5 4 Encoding Non-Quiet ·! 1贞 19
每个传输时间间隔内用户 B 任  User B in each transmission interval
A类 C类 A类 B类 C类 任意 数据的 AMR类型  Class A C class A class B class C class AMR type of arbitrary data
传输信道 1 2 3 1 2 3 5 4 承载用户数据的逻辑信道  Transport channel 1 2 3 1 2 3 5 4 Logical channel carrying user data
TM TM TM TM TM ΤΜ AM UM  TM TM TM TM TM ΤΜ AM UM
的模式  Mode
承载用户数据的逻辑信道  Logical channel carrying user data
4 4 4 4 4 4 2 6 的优先级 表格 3表示表格 2中的各传输信道的属性和参数, 特别是配置的传输格式 (TF)。 表格 3 4 4 4 4 4 4 2 6 priority Table 3 shows the attributes and parameters of each transport channel in Table 2, in particular the configured transport format (TF). Form 3
Figure imgf000021_0001
从表格 3中可以看出配置的传输格式支持的自适应多速率 (AMR) 话音编码器模式 为 12.2kbps、 7.4kbps和 4.75kbps, 无线接口功能单元中话音帧模式控制部分向 AMR话 音编码器发出的模式选择信号 9可包含上述 3种模式中的任意 2种, 例如 (12.2kbps, 4.75bps )。
Figure imgf000021_0001
It can be seen from Table 3 that the configured transport format supports adaptive multi-rate (AMR) voice coder modes of 12.2 kbps, 7.4 kbps and 4.75 kbps. The voice frame mode control part of the wireless interface function unit sends out to the AMR voice coder. The mode selection signal 9 may include any two of the above three modes, for example, (12.2 kbps, 4.75 bps).
表格 4给出所有传输格式组合的意义, 也就是传输格式组合用来传送的用户数据。 表格 4 传输信道 TrChl TrCh2 TrCh3 TrCh4 TrCh5 冠、  Table 4 gives the meaning of all combinations of transport formats, that is, the user data that the transport format combination is used to transmit. Table 4 Transport Channel TrChl TrCh2 TrCh3 TrCh4 TrCh5 crown,
TFI 0 0 0 0 0 不发送数据  TFI 0 0 0 0 0 does not send data
1 0 0 0 0 发送静默帧  1 0 0 0 0 Send silence frame
2 1 1 0 0 发送 12.2kbps模式话音数据 2 1 1 0 0 Send 12.2kbps mode voice data
3 2 0 0 0 发送 7.4kbps模式话音数据3 2 0 0 0 Send 7.4kbps mode voice data
4 3 0 0 0 发送 4.75kbps模式话音数据4 3 0 0 0 Send 4.75kbps mode voice data
0 0 0 1 0 发送分组数据 0 0 0 1 0 Send packet data
0 0 0 0 1 发送信令数据  0 0 0 0 1 Send signaling data
0 0 0 1 1 发送信令和分组数据 0 0 0 1 1 Send signaling and packet data
1 0 0 1 0 发送静默帧且发送分组数据1 0 0 1 0 Send silence frame and send packet data
1 0 0 0 1 发送静默帧且发送信令数据 发送静默帧且发送信令和分组数1 0 0 0 1 Sends a silence frame and sends signaling data. Sends a silence frame and sends signaling and number of packets.
1 0 0 1 1 1 0 0 1 1
据 发送 12.2kbps模式话音数据 According to Send voice data in 12.2kbps mode
2 1 1 1 0  2 1 1 1 0
且发送分组数据  And sending packet data
发送 12.2kbps模式话音数据 Send voice data in 12.2kbps mode
2 1 1 0 1 2 1 1 0 1
且发送 f言令数据  And send f command data
发送 12.2kbps模式话音数据 Send voice data in 12.2kbps mode
2 1 1 1 1 2 1 1 1 1
且发送信令和分组数据 发送 7.4kbps模式话音数据 And send signaling and packet data to send 7.4kbps mode voice data
3 2 0 1 0 3 2 0 1 0
且发送分组数据  And sending packet data
发送 7.4kbps模式话音数据 Send 7.4kbps mode voice data
3 2 0 0 1 3 2 0 0 1
且发送 f言令数据  And send f command data
发送 7.4kbps 模式话音数据 Send 7.4kbps mode voice data
3 2 0 1 1 3 2 0 1 1
且发送信令和分组数据 发送 4.75kbps模式话音数据 And send signaling and packet data to send 4.75kbps mode voice data
4 3 0 1 0 4 3 0 1 0
且发送分组数据  And sending packet data
发送 4.75kbps模式话音数据 Send 4.75kbps mode voice data
4 3 0 0 1 4 3 0 0 1
且发送信令数据  And sending signaling data
发送 4.75kbps模式话音数据 Send 4.75kbps mode voice data
4 3 0 1 1 4 3 0 1 1
且发送信令和分组数据 当模式选择信号 9包含的模式为 (12.2kbps, 4.75bps), 使得无线接口功能单元要处 理话音帧编码序列 2中的是 12.2kbps和 4.75bps的话音数据, 如果在需要调度到传输信 道信令数据 5和分组数据 6的输出中没有有效数据, 且传送格式组合 (2, 1, 1 , 0, 0) 又是有效组合, 无线接口功能单元可以发送模式为 12.2kbps的话音帧; 如果在需要调度 到传输信道信令数据 5中有有效数据需要传送, 分组数据中没有有 数据, 且由于物理 信道带宽的限制传送格式组合 (2, 1, 1, 0, 1 ) 是无效组合, 而传送格式组合 (4, 3, 0, 0, 1 )没有受物理信道的带宽限制是有效组合, 此时, 无线接口功能单元可以同时发 送模式为 4.75kbps的话音帧和信令。  And transmitting signaling and packet data when the mode selection signal 9 contains a mode (12.2 kbps, 4.75 bps), so that the radio interface functional unit is to process voice data of 12.2 kbps and 4.75 bps in the voice frame coding sequence 2, if There is no valid data in the output of the transport channel signaling data 5 and the packet data 6 to be scheduled, and the transport format combination (2, 1, 1 , 0, 0) is a valid combination, and the wireless interface functional unit can transmit the mode at 12.2 kbps. Voice frame; if there is valid data to be transmitted in the transmission channel signaling data 5, there is no data in the packet data, and the transmission format combination (2, 1, 1, 0, 1) is limited due to the limitation of the physical channel bandwidth. It is an invalid combination, and the transport format combination (4, 3, 0, 0, 1) is not effectively limited by the bandwidth limitation of the physical channel. At this time, the wireless interface functional unit can simultaneously transmit the voice frame and signaling with the mode of 4.75 kbps. .
当模式选择信号 9包含的模式为 (12.2kbps, 4.75bps), 使得无线接口功能单元要处 理话音帧编码序列 2中的是 12.2kbps和 4.75bps的话音数据, 如果 J¾时物理信道的配置 发生改变, 使得信道的带宽降低从而使传送格式组合(2, 1, 1, 0, 0) 变成无效组合, 而传送格式组合 (3, 2, 0, 0, 0)和 (4, 3, 0, 0, 0)没有受物理信道的带宽限制是 有效组合, 此时, 在仅有话音数据时无线接口功能单元发送模式为 4-75kbps的话音帧, 同时又向 AMR话音编码器发送包含 (7.4kbps, 4.75bps) 的模式选择信号 9, 在话音帧 编码序列 2中的变为 7.4kbps和 4.75bps的话音数据后, 在仅有话音数据时无线接口功能 单元可以发送模式为 7.4kbps的话音帧。 When the mode selection signal 9 contains a mode (12.2 kbps, 4.75 bps), the radio interface functional unit is to process voice data of 12.2 kbps and 4.75 bps in the voice frame coding sequence 2, and the configuration of the physical channel changes if J3⁄4 , causing the bandwidth of the channel to decrease such that the transport format combination (2, 1, 1, 0, 0) becomes an invalid combination, and the transport format combination (3, 2, 0, 0, 0) and (4, 3, 0, 0, 0) The bandwidth limit without the physical channel is a valid combination. At this time, the wireless interface function unit transmits a voice frame of 4-75 kbps mode when only voice data is available, and transmits the content to the AMR voice coder (7.4 kbps). Mode selection signal 9 of 4.75 bps), after changing to 7.4 kbps and 4.75 bps of voice data in the voice frame coding sequence 2, the wireless interface function only when voice data is available The unit can transmit a voice frame of mode 7.4 kbps.
图 3是模式选择信号 9包含的自适应多速率 (AMR)模式的数目为 3的編码模块的 处理示意图,数字话音信号 1同时向带话音激活检测的话音编码功能模块 100、 语音编码 模块 101和语音编码模块 102输出, 带话音激活检测的话音编码功能模块 100 同通用移 动通信***(UMTS )标准 TS26.071 (或 TS26.171 )的图 1话音处理功能示意(Overview of audio processing functions) 中的对应部分的结构基本相同, 区别是图 2中面出了输出 到语音编码模块的编码模式指示信号 17, 话音激活检测模块输出的是话音激 检测标志 (Voice Activity Detector flag) 10, 带话音激活检测的话音编码功能模块 100 语音编码 子模块输出话音编码帧 12, 语音编码模块 101和语音编码模块 102输出的分另 LI是话音编 码帧 19和 21, 话音编码帧 12、 19和 21的每帧比特数分别取决于编码模式指示信号 17、 18和 15, 编码模式指示信号 17、 18和 15是编码模式控制模块将模式选择信号 9中的 3 个模式分解成的 1 个模式 1 路的 3路输出。 不连续发送和操作模块输出自适应多速率 (AMR)帧类型信号 11到多路语音编码复用模块和语音编码模块,自适应多速率(AMR) 帧类型信号 11指示: 信息比特 14是否是有效的自适应多速率编码帧, 自适 多速率编 码帧的模式, 所述自适应多速率编码帧的模式为话音帧的模式或静默帧的模 ;, 当自适 应多速率编码帧的模式是静默帧时,信息比特 14是舒适噪声发送模块输出的静默检测帧 13。 语音编码模块 101和语音编码模块 102分别输出自适应多速率话音编码 19和 21 以及各自的自适应多速率 (AMR)帧类型信号 16和 20, 它们分别表示自适应多速率话 音编码帧 19和 21的模式。 多路语音编码复用模块将来自不连续发送和操作模块的信息 比特 14与自适应多速率话音编码帧 19和 21合并成话音帧编码序列 2, 合并 方法是: 当自适应多速率 (AMR) 帧类型信号 11指示信息比特 14无效时, 设置话音贞编码序列 2的模式指示为无效的话音帧编码序列; 当自适应多速率(AMR)帧类型信号 11指示信 息比特 14是静默帧时, 设置话音帧编码序列 2的模式指示为静默帧, 将信息 fc 特 14放 到话音帧编码序列 2的话音帧编码序列里; 当自适应多速率(AMR)帧类型信 11指示 信息比特 14是非静默帧时, 把信息比特 14、 19和 21—同放到话音帧编码序歹 ij 2的话音 帧编码序列里, 同时设置话音帧编码序列 2的模式指示为自适应多速率(AMR) 帧类型 信号 11、 16和 20所指示的模式。  3 is a schematic diagram of processing of an encoding module having a number of adaptive multi-rate (AMR) modes of 3 included in the mode selection signal 9. The digital voice signal 1 simultaneously transmits a voice encoding function module 100 with a voice activation detection, and a voice encoding module 101. And the speech encoding module 102 outputs, the voice encoding function module 100 with voice activation detection is in the Overview of audio processing functions of the Universal Mobile Telecommunications System (UMTS) standard TS26.071 (or TS 26.171). The structure of the corresponding part is basically the same, the difference is that the coding mode indication signal 17 outputted to the speech coding module is shown in FIG. 2, and the voice activity detection detection flag (Voice Activity Detector flag) 10 is outputted by the voice activation detection module, with voice activation. The detected speech encoding function module 100 has a speech encoding sub-module that outputs a speech encoding frame 12, and the speech encoding module 101 and the speech encoding module 102 output a separate LI that is a speech encoding frame 19 and 21, and each of the speech encoding frames 12, 19, and 21 The number of bits depends on the coding mode indication signals 17, 18 and 15, respectively. A signal indicative of formula 17, 18 and 15 are the encoding mode selected in the mode control module 9 3 3 mode signal into an output channel pattern 1. The discontinuous transmission and operation module outputs an adaptive multi-rate (AMR) frame type signal 11 to a multi-channel speech coding multiplexing module and a speech coding module, and an adaptive multi-rate (AMR) frame type signal 11 indicates: whether the information bit 14 is valid Adaptive multi-rate coded frame, mode of adaptive multi-rate coded frame, mode of the adaptive multi-rate coded frame is a mode of a voice frame or a mode of a silence frame; when the mode of the adaptive multi-rate coded frame is silent At the time of frame, the information bit 14 is the silence detection frame 13 output by the comfort noise transmitting module. The speech encoding module 101 and the speech encoding module 102 output adaptive multirate speech encodings 19 and 21 and respective adaptive multirate (AMR) frame type signals 16 and 20, respectively, which represent adaptive multirate speech encoding frames 19 and 21, respectively. Mode. The multiplexed speech coding multiplexing module combines the information bits 14 from the discontinuous transmission and operation modules with the adaptive multi-rate speech coding frames 19 and 21 into a speech frame coding sequence 2, the combining method being: when adaptive multi-rate (AMR) When the frame type signal 11 indicates that the information bit 14 is invalid, the mode of the voice 贞 code sequence 2 is set to be an invalid voice frame code sequence; when the adaptive multi-rate (AMR) frame type signal 11 indicates that the information bit 14 is a silence frame, The mode of the voice frame coding sequence 2 is indicated as a silence frame, and the information fc is placed in the voice frame coding sequence of the voice frame coding sequence 2; when the adaptive multi-rate (AMR) frame type signal 11 indicates that the information bit 14 is a non-silent frame At the same time, the information bits 14, 19 and 21 are placed in the voice frame coding sequence of the voice frame coding sequence 歹 ij 2 , and the mode indication of the voice frame coding sequence 2 is set as the adaptive multi-rate (AMR) frame type signal 11 , 16 and 20 indicate the mode.
实施例 2——移动台以间断方式传送自适应多速率(AMR)编码帧  Embodiment 2 - The mobile station transmits an adaptive multi-rate (AMR) coded frame in an intermittent manner
有缓存的移动台的生成和传送自适应多速率 (AMR)话音帧的方法的实施例参见图 4, 图 4和图 1的区别在于多出了先进先出存储器(FIFO)和用传输格式组合逸择先进先 出存储器读取控制部件替代了图 1中的传输格式组合选择部件。 编码模块的输出话音帧 编码序列 2到先进先出存储器(FIFO), 先进先出存储器(FIFO)缓存话音帧 码序列, 并输出存储状态标志 25,存储状态标志 25指示: 是否有未被读取的话音帧编码序列, 以 及这些未被读取的话音帧编码序列的数目。 传输格式组合选择先进先出存储 读取控制 部件输出读取命令 26, 读取命令 26使先进先出存储器 (FIFO) 输出活音帧编 ϊ¾序列 3。 为把话音的延迟控制在一定的限度之内, 传输格式组合选择先进先出存储器读取控 制部件可以检查存储状态标志 25给出的先进先出存储器(FIFO)中存储的话音 fe¾编码序 列的数目是否超过限定值, 当超过时即可确定有超时存储的话音帧编码序列, 然后以读 取命令 26将超时的话音帧编码序列从先进先出存储器(FIFO)中取出加以丢弃, 可由延 迟控制的程度而决定上述限定值。 An embodiment of a method for generating and transmitting an adaptive multi-rate (AMR) voice frame with a buffered mobile station is shown in FIG. 4. The difference between FIG. 4 and FIG. 1 is that the first-in first-out memory (FIFO) is combined with the transport format. The ED first-in memory read-out control component replaces the transport format combination selection component of FIG. The output voice frame coding sequence 2 of the encoding module is to the first in first out memory (FIFO), the first in first out memory (FIFO) buffers the voice frame code sequence, and outputs the storage status flag 25, and the storage status flag 25 indicates: whether there is any unread The voice frame coding sequence, and the number of these unread voice frame coding sequences. Transport format combination selects first-in-first-out storage read control The component outputs a read command 26, and the read command 26 causes the first in first out memory (FIFO) to output a live frame of the sequence 3 . In order to control the delay of the voice within a certain limit, the transport format combination selects the first in first out memory read control component to check the number of voice fe3⁄4 code sequences stored in the first in first out memory (FIFO) given by the storage status flag 25. Whether the limit value is exceeded, when it is exceeded, the voice frame coding sequence with timeout storage is determined, and then the time-out voice frame coding sequence is taken out from the first-in first-out memory (FIFO) by the read command 26, and can be discarded by delay control. The above limit value is determined to the extent.
表格 5表示在工作过程中的一个示例性的传输信道的配置, 给出各传输信道的属性 和参数, 特别是每一传输信道的传输格式标识(TFI)所对应的传输格式(TF) D传输时 间间隔(TTI)。  Table 5 shows the configuration of an exemplary transport channel in the course of operation, giving the attributes and parameters of each transport channel, in particular the transport format (TF) D transmission corresponding to the transport format identifier (TFI) of each transport channel. Time interval (TTI).
表格 5 Form 5
Figure imgf000024_0001
当模式选择信号 9包含的模式为 (12.2kbps, 4.75bps), 使得无线接口功能取元中存 储在先进先出存储器 (FIFO) 中的话音帧编码序列包含的是 12.2kbps和 4.75bps 模式的 话音数据, 如果在需要调度到传输信道信令数据 5中有有效数据需要传送, 分紅数据中 没有有效数据, 且由于物理信道带宽的限制传送格式组合(2, 1, 1, 0, 1 )是无 ¾组合, 无线接口功能单元在用传送格式组合 (0, 0, 0, 0, 1 )在一个 20毫秒的传输日 间间隔 内发送完信令数据后, 无线接口功能单元可以用 (4, 3, 0, 0, 0)在连续 2个 10毫秒 的传输时间间隔发送模式为 4.75kbps的被缓存的话音编码帧, 2个中的前一个^因抢占 而被延迟的, 后一个是未被延迟的, 这样, 受信令数据抢先而暂停发送的话音数据得到 了延迟的发送, 同时又不影响后续的话音帧的发送。 如果上述被抢占的是一个静默开始 帧, 可用 (5, 0, 0, 0, 0) 发送被缓存的这个静默开始帧的方法使之得到延迟 发送。 例如先进先出存储器(FIFO) 的超时的限定值被设定为 10个 20毫秒的时间长度的 移动台, 在发生切换时其无线接口功能单元无法发送无线帧, 当切换完成无线接口功能 单元可以发送无线帧时, 切换期间在先进先出存储器 (FIFO) 中存放了 8个话音帧编码 序列,这些话音帧编码序列同上一个示例一样是模式选择信号 9包含的模式为(12.2kbps, 4.75bps) 的结果, 无线接口功能单元可以用 (4, 3, 0, 0, 1 )发送一个模式为 4.75kbps 的自适应话音编码帧和从当前帧开始发送 16 (8+4+2+1+1 ) 个延迟自适应编码帧的信令 消息, 然后以 (4, 3, 0, 0, 0)在连续 15个 10毫秒的传输时间间隔发送切换期间缓存 的自适应话音编码帧和由此引起的在先进先出存储器(FIFO)内存放的时间超过 20毫秒 的延迟话音帧编码序列,之后就可以恢复模式为 12.2kbps的话音编码帧的 20毫秒一次的 发送了。
Figure imgf000024_0001
When the mode selection signal 9 contains a mode (12.2 kbps, 4.75 bps), the voice frame coding sequence stored in the first-in-first-out memory (FIFO) of the radio interface function fetcher contains voices of 12.2 kbps and 4.75 bps modes. Data, if there is valid data to be transmitted in the transmission channel signaling data 5, there is no valid data in the dividend data, and the transmission format combination (2, 1, 1, 0, 1) is none due to the limitation of the physical channel bandwidth. 3⁄4 combination, the wireless interface function unit can use the transmission interface format (0, 0, 0, 0, 1) after transmitting the signaling data in a 20 millisecond transmission day interval, the wireless interface function unit can use (4, 3 , 0, 0, 0) The buffered voice coded frame with the mode of 4.75 kbps is transmitted in two consecutive 10 millisecond transmission time intervals. The previous one of the two is delayed due to preemption, and the latter is undelayed. In this way, the voice data temporarily suspended by the signaling data is delayed to be transmitted without affecting the subsequent transmission of the voice frame. If the above preemption is a silent start frame, the method of transmitting the silenced start frame can be sent (5, 0, 0, 0, 0) to delay transmission. For example, the time limit of the first-in first-out memory (FIFO) is set to 10 mobile stations with a length of 20 milliseconds. When the switchover occurs, the wireless interface function unit cannot send the wireless frame. When the switch completes the wireless interface function unit, When transmitting a radio frame, eight voice frame coding sequences are stored in the first in first out memory (FIFO) during the switching. These voice frame coding sequences are the same as the previous example. The mode selection signal 9 contains the mode (12.2 kbps, 4.75 bps). As a result, the wireless interface function unit can transmit (4, 3, 0, 0, 1) an adaptive voice coded frame of mode 4.75 kbps and transmit 16 (8+4+2+1+1) from the current frame. Delaying the adaptively encoded framed signaling message, and then transmitting (4, 3, 0, 0, 0) the adaptive voice coded frame buffered during the switching for 15 consecutive 10 millisecond transmission time intervals and the resulting The delayed voice frame coding sequence stored in the FIFO for more than 20 milliseconds can be recovered after 20 milliseconds of the voice coded frame of mode 12.2 kbps.
将图 4中的话音帧模式控制部分和传输格式组合选择先进先出存储器读取控制部分 中的传输格式组合选择放到中央处理器上去实现, 将传输格式组合选择先进先出存储器 读取控制部分中的先进先出存储器读取控制放到信道编码器中去实现就形成了图 5所示 的移动台中缓存的接口的实施例。  The combination of the voice frame mode control portion and the transport format combination in FIG. 4 selects the transport format combination in the FIFO read control portion to be implemented on the central processing unit, and selects the transport format combination to select the FIFO memory read control portion. The FIFO read control is placed in the channel encoder to implement an embodiment of the interface buffered in the mobile station shown in FIG.
参见图 5, 先进先出 (FIFO) 缓存处于自适应多速率 (AMR) 编码器和信道编码器 之间, 自适应多速率(AMR)编码器每 20毫秒向先进先出 (FIFO) 缓存输出一个话音 帧编码序列, 在图中的先进先出 (FIFO) 缓存用来存储话音帧编码序列, 中央处理器通 过图中示出的接口 29读出待处理的话音帧编码序列的类型, 中央处理器可通过图中示出 的接口 28向自适应多速率(AMR)编码器发送包含多个自适应多速率(AMR)模式的 编码命令。  Referring to Figure 5, the first in first out (FIFO) buffer is between the adaptive multi-rate (AMR) encoder and the channel encoder, and the adaptive multi-rate (AMR) encoder outputs one to the first in first out (FIFO) buffer every 20 milliseconds. The voice frame coding sequence, the first in first out (FIFO) buffer in the figure is used to store the voice frame coding sequence, and the central processor reads out the type of the voice frame coding sequence to be processed through the interface 29 shown in the figure, the central processor An encoding command comprising a plurality of adaptive multi-rate (AMR) modes can be transmitted to the adaptive multi-rate (AMR) encoder via the interface 28 shown in the figure.
表格 6给出了一个例子,这个例子说明了在表格 6中的 0到 80毫秒的这段时间内根 据先进先出 (FIFO) 缓存中被写入的话音帧编码序列所做的传输信道格式组合 (TFC) 选择, 以及如何根据传输信道格式组合 (TFC)选择的结果来读取先进先出 (FIFO) 缓 存中的话音帧编码序列中的自适应多速率 (AMR)编码帧。  Table 6 gives an example of a combination of transport channel formats based on the sequence of voice frame codes written in the first-in, first-out (FIFO) buffer during the period from 0 to 80 milliseconds in Table 6. (TFC) Select, and how to read an adaptive multi-rate (AMR) encoded frame in a sequence of voice frame encodings in a first in first out (FIFO) buffer based on the results of a Transport Channel Format Combination (TFC) selection.
表格 6 时间 先 进 先 出 待处理的话音帧编 选择的传输格式组 向信道编码器 Table 6 Time First In First Out Outgoing Voice Frame Editing Selected Transmission Format Group To Channel Encoder
(毫秒) (FIFO)缓存 码序列及其类型 合中用于话音的格 输出的自适应 中的话音帧编 式及传输时间间隔 多速率 (AMR)帧 码序列 (TTI) (milliseconds) (FIFO) buffer code sequence and its type. The voice frame format and transmission time interval in the adaptive output of the speech. Multi-rate (AMR) frame code sequence (TTI)
0 305 305,模式 1+模式 7 (81,103,60),20毫秒 模式 7  0 305 305, mode 1+ mode 7 (81,103,60), 20 ms mode 7
15 304  15 304
20 304 304,模式 1+模式 7 无 无  20 304 304, mode 1+ mode 7 none
35 304, 303  35 304, 303
40 304, 303 304,模式 1+模式 7 (49, 54), 10毫秒 模式 1 50 303 303,模式 1+模式 2 (55, 63), 20毫秒 模式 240 304, 303 304, Mode 1 + Mode 7 (49, 54), 10 ms mode 1 50 303 303, mode 1 + mode 2 (55, 63), 20 ms mode 2
55 302 55 302
60 302 302,模式 1+模式 2 不适用 不适用  60 302 302, Mode 1+ Mode 2 Not applicable Not applicable
70 302 302,模式 1+模式 2 (55, 63), 10毫秒 模式 2  70 302 302, mode 1+ mode 2 (55, 63), 10 ms mode 2
75 301 301 ,模式 1+模式 7  75 301 301, mode 1+ mode 7
80 301 305,模式 1+模式 7 (81,103,60),20毫秒 模式 7 图 6表示的 0毫秒时先进先出(FIFO)缓存中存储的是话音帧编码序列 305, 表格 6 中示出,此时中央处理器通过图中示出的接口 29读出的待处理的话音帧编码序列是有效 的话音帧编码序列 305, 其类型为模式 1+模式 7, 中央处理器上进行的传输信道格式组 合(TFC)选择的结果是一个传输时间间隔 (TTI) 为 20毫秒的组合, 其中包含用于话 音的传输信道的 3个格式, 1个格式为 1 X 89比特, 另一个为 1 X 103比特, 还有一个是 1 X 60 比特。 该结果实际上指定要将话音帧编码序列 305 中的模式 7 的自适应多速率 (AMR) 帧放到传输信道上, 图 7中的 501就是该模式 7的自适应多速率(AMR) 帧, 具体的操作是这样的, 中央处理器通过接口 27向信道编码器发出根据包含 1 X 89比特、 1 X 103比特和 1 X 60比特这 3个话音传输格式的传输格式组合进行编码的指令, 这将使 其发出读取先进先出(FIFO)缓存话音帧编码序列 305中模式 7的自适应多速率(AMR) 编码帧 501的操作, 该操作将话音帧编码序列 305移出先进先出 (FIFO)缓存。  80 301 305, Mode 1 + Mode 7 (81, 103, 60), 20 millisecond mode 7 Figure 6 shows the 0 millisecond time in the first in first out (FIFO) buffer stored in the voice frame coding sequence 305, as shown in Table 6, this The sequence of the speech frame to be processed read by the central processing unit through the interface 29 shown in the figure is a valid voice frame coding sequence 305 of the type 1 + mode 7, the combination of the transmission channel format on the central processing unit. The result of the (TFC) selection is a combination of a transmission time interval (TTI) of 20 milliseconds, which contains three formats for the voice transmission channel, one format being 1 X 89 bits and the other being 1 X 103 bits. There is also a 1 X 60 bit. The result actually specifies that the adaptive multi-rate (AMR) frame of mode 7 in the voice frame coding sequence 305 is to be placed on the transmission channel, 501 in Figure 7 is the adaptive multi-rate (AMR) frame of mode 7. The specific operation is such that the central processor issues an instruction to the channel coder via the interface 27 to encode according to a combination of transmission formats including 3 X 89 bits, 1 X 103 bits and 1 X 60 bits. It will be operative to read the adaptive multi-rate (AMR) encoded frame 501 of mode 7 in the first in first out (FIFO) buffered speech frame encoding sequence 305, which shifts the speech frame encoding sequence 305 out of the first in first out (FIFO) Cache.
表格 6所示在 20毫秒时, 中央处理器通过图中示出的接口 29读出待处理的话音帧 编码序列是有效的话音帧编码序列 304, 其类型为模式 1+模式 7, 传输信道格式组合 (TFC) 选择的结果是没有能传输话音的传输格式组合, 有很多原因都会导致这样的结 果(物理信道变化、 高优先级逻辑信道以及切换), 所以此时不从先进先出(FIFO)缓存 读取话音帧编码序列 304。  Table 20 shows that at 20 milliseconds, the central processor reads out the voice frame coding sequence to be processed by the interface 29 shown in the figure as a valid voice frame coding sequence 304 of the type 1 + mode 7, transmission channel format. The result of the combination (TFC) selection is that there is no transport format combination that can transmit voice. There are many reasons for this (physical channel change, high priority logical channel, and handover), so this is not the first in first out (FIFO). The cache reads the voice frame encoding sequence 304.
图 8表示 35毫秒时先进先出 (FIFO)缓存中存储的是话音帧编码序列 304, 同时自 适应多速率 (AMR)编码器向先进先出 (FIFO)缓存输出话音帧编码序列 303, 话音帧 编码序列 303的类型是模式 1+模式 2, 比起上一个话音帧编码序列 304来发生了改变, 即中央处理器的编码命令中的模式的改变在话音帧编码序列 303中反映出来了。  Figure 8 shows a voice frame encoding sequence 304 stored in a first in first out (FIFO) buffer at 35 milliseconds, while an adaptive multi-rate (AMR) encoder outputs a voice frame encoding sequence 303, a voice frame to a first in first out (FIFO) buffer. The type of the coding sequence 303 is mode 1 + mode 2, which is changed compared to the previous speech frame coding sequence 304, i.e., the mode change in the coding command of the central processor is reflected in the speech frame coding sequence 303.
图 9表示 40毫秒时先进先出 (FIFO) 缓存中存储的是话音帧编码序列 304和 303, 表格 6中示出, 此时因为先进先出 (FIFO) 缓存中存储的话音帧编码序列的数量超过了' 1,优选 10毫秒的传输时间间隔(TTI)的传输格式组合, 此时传输信道格式组合(TFC) 选择的结果是一个传输时间间隔 (ΤΉ) 为 10毫秒的组合, 其中包含用于话音的传输信 道的 2个格式, 1个格式为 1 X49比特, 另一个为 1 X 54比特, 该结果实际上指定要将话 音帧编码序列 304中的模式 1的自适应多速率(AMR) 帧放到传输信道上。 图 10中的 401就是该模式 1的自适应多速率 (AMR) 帧, 具体的操作是这样的, 中央处理器通过 接口 27向信道编码器发出根据包含 1 X49比特和 1 X 54比特这 2个话音传输格式的传输 格式组合进行编码的指令,这将使其发出读取先进先出(FIFO)缓存话音帧编码序列 304 中模式 1的自适应多速率(AMR) 编码帧 401的操作, 该操作将话音帧编码序列 304,移 出先进先出 (FIFO) 缓存。 Figure 9 shows the voice frame encoding sequences 304 and 303 stored in the first in first out (FIFO) buffer at 40 milliseconds, as shown in Table 6, at this time because of the number of voice frame encoding sequences stored in the first in first out (FIFO) buffer. A transmission format combination exceeding '1, preferably 10 milliseconds, of transmission time interval (TTI), at which point the result of the Transport Channel Format Combination (TFC) selection is a combination of a transmission time interval (ΤΉ) of 10 milliseconds, which is included for The two formats of the voice transmission channel, one format being 1 X49 bits and the other being 1 X 54 bits, the result actually specifying the adaptive multi-rate (AMR) frame of mode 1 to be encoded in the voice frame coding sequence 304. Put it on the transmission channel. 401 in FIG. 10 is the adaptive multi-rate (AMR) frame of the mode 1, and the specific operation is such that the central processor sends the channel encoder through the interface 27 according to the two of the X X49 bits and the 1 X 54 bits. Voice transmission format transmission The format combines the encoded instructions, which will cause it to issue an operation of reading the adaptive multi-rate (AMR) encoded frame 401 of mode 1 in the first in first out (FIFO) buffered speech frame encoding sequence 304, which will sequence the speech frame encoding 304, remove the first in first out (FIFO) cache.
由于表格 6示出的 55毫秒时写入的话音帧编码序列 302直到 70毫秒时才被读出, 写入和读取时间间隔超出 10毫秒, 由于传输信道格式组合 (TFC)选择可用于缩短所述 写入和读取时间间隔, 所以 70毫秒时传输信道格式组合 (TFC)选择的结果是选用 10 毫秒传输时间间隔 (TTD 的传输格式组合, 即表格中所示的 "(55, 63), 10毫秒", 这 将使 80毫秒时的读取的操作与 75毫秒的写入操作这对操作的时间间隔与上次的那对操 作相比有了较大的减少。  Since the voice frame coding sequence 302 written at 55 milliseconds as shown in Table 6 is not read until 70 milliseconds, the write and read time intervals exceed 10 milliseconds, since the Transport Channel Format Combination (TFC) selection can be used to shorten the The write and read time intervals are described, so the result of the Transmission Channel Format Combination (TFC) selection at 70 milliseconds is the 10 ms transmission time interval (the combination of the transport format of the TTD, ie the table shown in the table "(55, 63), 10 milliseconds", this will result in a significant reduction in the time interval between the read operation at 80 milliseconds and the write operation at 75 milliseconds compared to the last pair of operations.
实施例 3——移动台延迟自适应多速率(AMR)译码  Embodiment 3 - Mobile Station Delay Adaptive Multi-Rate (AMR) Decoding
参见图 11, 移动台的接收信源控制速率处理器 (Rx SCR handler)通过先进先出存 储器(FIFO)与无线接口功能单元相联系, 32、 33和 34分别是先进先出存储器(FIFO) 向移动台的接收信源控制速率处理器(Rx SCR handler) 输出的每个接收帧的信息比特、 模式指示和接收类型, 37、 38和 39分别是无线接口功能单元向先进先出存储器(FIFO) 输出的每个接收帧的信息比特、 模式指示和接收类型, 模式指示给出接收帧的自适应多 速率(AMR模式, 接收类型如表格 7所示。先进先出存储器(FIFO)将无线接口功能单 元送出的每一个自适应多速率 (AMR) 接收帧按序缓存, 缓存时将上述的信息比特、 模 式指示和接收类型一起保存。在缓存状态标志 30指示: 是否有未被读取的自适应多速率 (AMR)接收帧, 以及这些未被读取的自适应多速率(AMR)接收帧的数目。 当一个话 音呼叫开始时接收信源控制速率处理器(R SCR handler)不立即去将收到的未被读取的 接收帧读取并译码, 而是当先进先出存储器(FIFO) 中存放的未被读取的接收帧的数目 达到了为先进先出存储器(FIFO) 这个缓存区设置的下限值时, 移动台的接收信源控制 速率处理器(SCRhandler)才开始每 20毫秒从中读取一个自适应多速率编码帧, 读取的 方法是向先进先出存储器(FIFO)发出读取命令 31。 先进先出存储器(FIFO)输出接收 丢失状态标志 35, 它给出移动台在语音 (SPEECH)模式下主动添加的接收丢失帧(类 型为 NO— DATA— SPEECH)的个数。 当接收丢失状态标志 35给出的值表明先进先出存储 器(FIFO) 中存放的在语音 (SPEECH) 模式下主动添加的接收丢失帧数目已经超过了 规定的限制值时,无线接口功能单元不再往先进先出存储器(FIFO)输出语音(SPEECH) 模式下添加的接收丢失帧。如果无线接口功能单元以每 20毫秒超过 1帧的速率向先进先 出存储器(FIFO)输出接收帧时, 无线接口功能单元向先进先出存储器(FIFO)发送删 除命令 36将先进先出存储器(FIFO) 中存放的在语音(SPEECH)模式下主动添加的接 收丢失帧删除。  Referring to Figure 11, the mobile station's receive source control rate processor (Rx SCR handler) is associated with the radio interface functional unit through a first in first out memory (FIFO), 32, 33 and 34 are first in first out memory (FIFO) directions, respectively. The receiving bit of the mobile station controls the information bits, mode indication and reception type of each received frame output by the Rx SCR handler, 37, 38 and 39 are respectively the wireless interface function unit to the first in first out memory (FIFO) The information bits, mode indication and reception type of each received frame are output. The mode indication gives the adaptive multi-rate of the received frame (AMR mode, the reception type is shown in Table 7. The first in first out memory (FIFO) will be the wireless interface function. Each adaptive multi-rate (AMR) receive frame sent by the unit is sequentially cached, and the above information bits, mode indication and reception type are saved together during buffering. The cache status flag 30 indicates: whether there is an unread adaptive Multi-rate (AMR) received frames, and the number of these unread adaptive multi-rate (AMR) received frames. When a voice call The R SCR handler does not immediately read and decode the received unread received frame, but is not read in the first in first out memory (FIFO). When the number of received frames reaches the lower limit set for the first-in-first-out memory (FIFO) buffer, the mobile station's receive source control rate processor (SCRhandler) begins to read an adaptive from every 20 milliseconds. The multi-rate encoded frame is read by issuing a read command 31 to the first in first out memory (FIFO). The first in first out memory (FIFO) output receives the lost status flag 35, which gives the mobile station in speech (SPEECH) mode. The number of actively added receive lost frames (of type NO_DATA_SPEECH). The value given by Receive Loss Status Flag 35 indicates that the value added in voice (SPEECH) mode stored in the First In First Out Memory (FIFO) is actively added. When the number of received lost frames has exceeded the specified limit, the wireless interface function unit no longer adds the received loss to the first-in-first-out memory (FIFO) output voice (SPEECH) mode. Frame loss. If the wireless interface function unit outputs a receive frame to the first in first out memory (FIFO) at a rate of more than 1 frame every 20 milliseconds, the wireless interface function unit sends a delete command 36 to the first in first out memory (FIFO) to be first in, first out. Receive lost frames deleted in voice (SPEECH) mode stored in the memory (FIFO).
表格 8给出了一个移动台延迟译码的一系列对接收帧的操作, 在这个示例中, 先进 先出存储器(FIFO)这个缓存区的下限值为 4, 它的在语音(SPEECH)模式下添加接收 丢失帧数目的规定限制值为 2。表格中列出了图 11中的各单元在第一个 20毫秒到第八个 20毫秒时间段内对所接收到的帧的处理。 Table 8 shows a series of operations on the received frame delayed by the mobile station. In this example, the first-in-first-out memory (FIFO) buffer has a lower limit of 4, and its speech (SPEECH) mode. Add and receive The specified limit value for the number of lost frames is 2. The processing of the received frames in the first 20 milliseconds to the eighth 20 millisecond time period of each unit in FIG. 11 is listed in the table.
表格 7  Form 7
Figure imgf000028_0001
Figure imgf000028_0001
20 毫秒 无线接口功能单元 先进先出 AMR SCR 注释  20 ms Wireless Interface Function Unit First In First Out AMR SCR Comments
单位的次 接收到的帧及类型 存储器中 控制器读  Unit of the received frame and type of memory in the controller read
序 缓存的帧 入的帧  Cached framed frame
1 第一帧 第一帧 无 初始 时移动 台 处于  1 first frame first frame no initial time mobile station is in
SPEECH—GOOD 空 SPEECH模式, 无线接口 空 功能单元把第一帧写入先 空 进先出存储器, Rx SCR handler不读取帧  SPEECH—GOOD Empty SPEECH mode, wireless interface The empty function unit writes the first frame to the first-in first-out memory, and the Rx SCR handler does not read the frame.
2 第二帧 第一帧 无 移动台仍处于 SPEECH模  2 Second frame First frame None Mobile station is still in SPEECH mode
SPEECH— GOOD 第二帧 式, 无线接口功能单元把 空 第二帧写入先进先出存储 空 器, Rx SCR handler仍不读 取帧  SPEECH—GOOD second frame, the wireless interface function unit writes the empty second frame to the FIFO memory, and the Rx SCR handler still does not read the frame.
3 第三帧 第一帧 无 移动台仍处于 SPEECH模  3 Third frame First frame None Mobile station is still in SPEECH mode
SPEECH— GOOD 第二帧 式, 无线接口功能单元把 第三帧 第三帧写入先进先出存储 空 器, Rx SCR handler仍不读 取帧 SPEECH—GOOD second frame, the wireless interface function unit writes the third frame of the third frame to the FIFO storage. Empty, Rx SCR handler still does not read frames
第四帧 第二帧 第一帧 移动台仍处于 SPEECH模Fourth frame second frame first frame mobile station is still in SPEECH mode
NO—DATA— SPEEC 第三帧 式, 无线接口功能单元把 H 第四帧 第四帧写入, 先进先出存 空 储器中的个数达到 4, Rx 空 SCR handler开始每 20毫 秒一个地读取帧 NO—DATA—SPEEC third frame, the wireless interface function unit writes the fourth frame of H fourth frame, the number of the first-in first-out memory is up to 4, and the Rx empty SCR handler starts reading every 20 milliseconds. Frame fetch
第五帧 第三帧 第二帧 移动台仍处于 SPEECH模Fifth frame third frame second frame mobile station is still in SPEECH mode
SPEECH— GOOD 第五帧 式, 无线接口功能单元把 第六帧 第六帧 第五和第六帧写入, 删除SPEECH—GOOD fifth frame, the wireless interface function unit writes the sixth frame, the sixth frame, the fifth and sixth frames, deletes
SPEECH— GOOD 空 先进先出存储器中的第四 空 帧一 SPEECH模式下添加 的接收丢失帧 SPEECH—GOOD Empty Fourth Empty Frame in FIFO Memory Receive Lost Frame Added in SPEECH Mode
第七帧 第五帧 第三帧 移动台仍处于 SPEECH模Seventh frame, fifth frame, third frame, mobile station is still in SPEECH mode
NO— DATA— SPEEC 第六帧 式, 无线接口功能单元把 H 第七帧 第七帧写入 NO—DATA—SPEEC sixth frame, the wireless interface function unit writes the seventh frame of H seventh frame
 Empty
第八帧 第六帧 第五帧 移动台仍处于 SPEECH模The eighth frame, the sixth frame, the fifth frame, the mobile station is still in the SPEECH mode.
NO—DATA—SPEEC 第七帧 式, 无线接口功能单元把 H 第八帧 第八帧写入 NO—DATA—SPEEC seventh frame, the wireless interface function unit writes the eighth frame of H eighth frame
 Empty
第九帧 第七帧 第六帧 移动台仍处于 SPEECH模Ninth frame seventh frame sixth frame mobile station is still in SPEECH mode
NO—DATA—SPEEC 第八帧 式, 先进先出存储器中已 H 空 经有了两个 SPEECH模式 下添加的接收丢失帧, 不 再写入 SPEECH模式下添 加的接收丢失帧第九帧 第十帧 第十一帧 第十帧 无线接口功能单元把第十NO-DATA-SPEEC The eighth frame type, the first-in first-out memory has been empty. There are two received loss frames added in the SPEECH mode, and the ninth frame of the ninth frame added in the SPEECH mode is no longer written. The eleventh frame tenth frame wireless interface function unit puts the tenth
SPEECH— GOOD 第八帧 和第十一帧写入到 第十一帧 空 SPEECH模式下添加的接 SPEECH— GOOD 收丢失帧前面, 删除先进 先出存储器中的第七帧一 SPEECH模式下添加的接 收丢失帧, R SCR handler 读取第十帧 SPEECH—GOOD The eighth and eleventh frames are written to the eleventh frame in the empty SPEECH mode. SPEECH—GOOD receives the lost frame in front of the frame, deletes the seventh frame in the FIFO memory, adds the received loss frame in SPEECH mode, and the R SCR handler reads the tenth frame.
10 第十二帧 第十二帧 第十一帧 无线接口功能单元把第十  10 twelfth frame twelfth frame eleventh frame wireless interface function unit put tenth
SPEECH—GOOD 第十三帧 二和第十三帧写入, 删除 第十三帧 空 先进先出存储器中的第八 SPEECH—GOOD 帧一 SPEECH模式下添加 的接收丢失帧, Rx SCR handler读取第 ^—帧 SPEECH-GOOD writes the thirteenth frame two and thirteenth frames, deletes the eighth SPEECH-GOOD frame in the thirteenth frame empty first-in first-out memory, the received lost frame added in the SPEECH mode, and the Rx SCR handler reads the first ^—frame
11 第十四帧 第十三帧 第十二帧 无线接口功能单元把第十 11 Fourteenth frame Thirteenth frame Twelfth frame Wireless interface function unit puts the tenth
SPEECH— GOOD 第十四帧 四和第十五帧写入, 因为 第十五帧 第十五帧 前面没有 SPEECH模式下 SPEECH GOOD 空 添加的接收丢失帧, 所以 没有删除操作, Rx SCR handler读取第十二帧 图 12给出一个实现上述延迟译码方法的框图, 图 11中的无线接口功能被放到信道 解码器和中央处理器中去实现, 移动台接收信源控制速率处理器被放到变速率译码器和 中央处理器中去实现。 先进先出 (FIFO) 缓存处于变速率译码器和信道解码器之间, 信 道解码器按 10毫秒帧定时进行解码操作, 中央处理器通过接口 54接收信道解码器发送 的生成自适应多速率 (AMR) 帧消息, 中央处理器可以通过接口 54 向信道解码器发出 将自适应多速率(AMR)帧输出到先进先出 (FIFO)缓存的指令; 中央处理器可通过接 口 53读取先进先出 (FIFO) 缓存中存储的自适应多速率 (AMR) 编码帧队列的长度和 这些编码帧的类型, 每隔 20毫秒中央处理器通过接口 53向先进先出 (FIFO)缓存发出 删除其中指定帧的命令; 中央处理器可通过接口 58向变速率译码器发出读取自适应多速 率(AMR)帧的指令。  SPEECH—GOOD is written in the fourteenth frame and the fifteenth frame. Because the fifteenth frame is not preceded by the SPEECH GOOD empty received loss frame in SPEECH mode, there is no delete operation, and the Rx SCR handler reads the first. Twelve Frames Figure 12 shows a block diagram of the above-described delay decoding method. The radio interface function in Figure 11 is implemented in the channel decoder and the central processor. The mobile station receives the source control rate processor and puts it on. The variable rate decoder and the central processor are implemented. The first in first out (FIFO) buffer is between the variable rate decoder and the channel decoder, the channel decoder performs decoding operations at 10 millisecond frame timing, and the central processor receives the adaptive multi-rate generated by the channel decoder via interface 54 ( AMR) frame message, the central processor can send an instruction to the channel decoder to output an adaptive multi-rate (AMR) frame to the first in first out (FIFO) buffer via interface 54; the central processor can read the first in first out through interface 53 (FIFO) The length of the adaptive multi-rate (AMR) encoded frame queue stored in the buffer and the type of these encoded frames. Every 20 milliseconds, the central processor issues a delete to the first in first out (FIFO) buffer through interface 53. The central processor may issue an instruction to read the adaptive multi-rate (AMR) frame to the variable rate decoder via interface 58.
参见图 13, 由于信道解码器在 2个 10毫秒帧定时期间没有收到包含自适应多速率 See Figure 13, since the channel decoder did not receive an adaptive multi-rate during two 10 ms frame timings.
(AMR)帧的无线帧, 通过接口 54向中央处理器发送无自适应多速率(AMR)帧指示, 中央处理器通过接口 54 向信道解码器发出输出无数据 (N0JMTA) 帧 52到先进先出The radio frame of the (AMR) frame transmits an unadaptive multi-rate (AMR) frame indication to the central processor via interface 54, and the central processor sends an output no data (N0JMTA) frame 52 to the channel decoder via interface 54 to first in first out.
(FIFO)缓存的指令, 并且, 此时先进先出 (FIFO)中有 2个自适应多速率(AMR)帧 55和 56, 中央处理器通过接口 58向变速率译码器发出将自适应多速率(AMR)帧从先 进先出 (FIFO) 缓存读出到自适应多速率(AMR)译码器的指令, 如图 14所示, 收到 该指令后自适应多速率(AMR)译码器从先进先出 (FIFO)缓存中读出的自适应多速率 (AMR)编码帧 56。 (FIFO) cached instructions, and, at this time, there are two adaptive multi-rate (AMR) frames 55 and 56 in the first in first out (FIFO), and the central processor sends more adaptive to the rate decoder through interface 58. Rate (AMR) frame from the first The first in first out (FIFO) buffer reads the instructions to the adaptive multi-rate (AMR) decoder, as shown in Figure 14, after receiving the instruction, the adaptive multi-rate (AMR) decoder is from the first in first out (FIFO) An adaptive multi-rate (AMR) encoded frame 56 read out of the buffer.
接下来的 2个 10毫秒帧定时信道解码器解码了 2个自适应多速率 (AMR)帧 51和 50, 并在每个自适应多速率 (AMR) 帧解码完成后通过接口 54 向中央处理器发送生成 自适应多速率 (AMR) 帧消息, 每次收到该种消息后中央处理器通过接口 54 向信道解 码器发出将自适应多速率 (AMR)帧输出到先进先出 (FIFO) 缓存的指令, 图 15示出 了收到这 2个自适应多速率(AMR)帧之后的先进先出(FIFO)缓存中存储的帧的情形。 中央处理器通过接口 53 获取先进先出 (FIFO) 缓存存储了无数据 (NO— DATA) 帧 52 的信息, 发出删除该无数据 (NO— DATA) 帧的指令, 执行该指令后的先进先出 (FIFO) 缓存中存储的帧的情形如图 16所示。  The next two 10 millisecond frame timing channel decoders decode two adaptive multi-rate (AMR) frames 51 and 50 and pass through interface 54 to the central processor after each adaptive multi-rate (AMR) frame decoding is completed. The transmit generates an adaptive multi-rate (AMR) frame message, and each time the message is received, the central processor sends an adaptive multi-rate (AMR) frame to the first in first out (FIFO) buffer via the interface 54 to the channel decoder. Instruction, Figure 15 shows the situation of frames stored in a first in first out (FIFO) buffer after receiving these two adaptive multi-rate (AMR) frames. The central processing unit acquires a first in first out (FIFO) buffer through the interface 53 to store information of the no data (NO-DATA) frame 52, issues an instruction to delete the no data (NO-DATA) frame, and executes the first in first out after the instruction. (FIFO) The situation of the frames stored in the buffer is as shown in FIG. 16.
实施例 4——网络侧生成和传送自适应多速率(AMR) 编码帧  Embodiment 4 - Network Side Generation and Transmission of Adaptive Multi-Rate (AMR) Coded Frames
参见图 17, 如图所示, 码型变换器 (TC) 对话音编码信号 41作编码转换, 产生话 音帧编码序列 42, 每个话音帧编码序列包含若干个自适应多速率(AMR)话音编码帧或 1个自适应多速率(AMR)静默帧或 1个无数据(NO— DATA)类型帧, 话音帧编码序列 42中的每一帧不仅包括自适应多速率(AMR)核心帧的信息比特, 还包括编码器循环冗 余校验(CRC:)、 帧类型、 质量指示和模式指示。 码型变换器 (TC) 产生的话音帧编码 序列 42经 Iu接口功能单元处理后先形成 Iu用户平面帧 43输入到无线网络控制器 (RNC), 再经 Iu接口功能单元处理后还原成话音帧编码序列 42,然后输出到传输格式组合选择单 元, 话音帧编码序列 42 中每帧的帧类型指示该帧是话音帧还是静默帧还是无数据 Referring to Figure 17, as shown, a transcoder (TC) speech encoded signal 41 is transcoded to produce a speech frame encoding sequence 42, each speech frame encoding sequence comprising a number of adaptive multi-rate (AMR) speech encodings. A frame or an adaptive multi-rate (AMR) silence frame or a no-data (NO-DATA) type frame, each frame in the voice frame coding sequence 42 includes not only information bits of an adaptive multi-rate (AMR) core frame. Also includes encoder cyclic redundancy check (CRC:), frame type, quality indication, and mode indication. The voice frame coding sequence 42 generated by the pattern converter (TC) is processed by the Iu interface function unit to form an Iu user plane frame 43 and input to the radio network controller (RNC), and then restored to a voice frame by the Iu interface function unit. The code sequence 42 is then output to the transport format combination selection unit, and the frame type of each frame in the voice frame coding sequence 42 indicates whether the frame is a voice frame or a silence frame or no data.
(NO_DATA)类型帧, 话音帧编码序列 42中每帧的模式指示给出了该帧的编码模式。 无线网络控制器(RNC) 中的传输格式组合选择模块以话音帧编码序列 42、 分组业务数 据 44和信令数据 45为输入, 传输格式组合选择模块选定一传输格式组合, 根据该传输 格式组合确定传送话音的传输格式以及对应的输出传输块。话音帧编码序列 42中的帧是 话音帧或静默帧而不是无数据(NO— DATA)类型帧时, 其对应的传输块是自适应多速率(NO_DATA) type frame, the mode indication of each frame in the voice frame coding sequence 42 gives the coding mode of the frame. The transport format combination selection module in the radio network controller (RNC) takes as input a voice frame coding sequence 42, packet service data 44, and signaling data 45, and the transport format combination selection module selects a transport format combination, and combines according to the transport format. Determine the transport format of the transmitted voice and the corresponding output transport block. When the frame in the voice frame coding sequence 42 is a voice frame or a silence frame instead of a no-data (NO-DATA) type frame, the corresponding transport block is an adaptive multi-rate.
(AMR) 编码帧 46, 这个自适应多速率 (AMR) 编码帧 46是话音帧编码序列 42中的 若干个自适应多速率 (AMR)话音帧中的 1个或者是其中的 1个静默帧, 分组业务数据 44和信令数据 45的对应传输块分别是分组数据传输块 48和信令数据传输块 47。无线网 络控制器(RNC) 的其它单元将话音数据传输块 46、 分组数据传输块 48和信令数据传 输块 47 映射到物理信道上进行发送。 无线网络控制器(RNC) 向核心网的媒体网关(AMR) coded frame 46, this adaptive multi-rate (AMR) coded frame 46 is one of a number of adaptive multi-rate (AMR) voice frames in the voice frame coding sequence 42 or one of the silence frames, The corresponding transport blocks of the packet service data 44 and the signaling data 45 are a packet data transport block 48 and a signaling data transport block 47, respectively. The other units of the Radio Network Controller (RNC) map the voice data transport block 46, the packet data transport block 48, and the signaling data transport block 47 onto the physical channel for transmission. Radio Network Controller (RNC) Media Gateway to the Core Network
(MGW)输出发送核心网-无线接入网接口 (Iu)用户平面速率控制帧 49, 核心网 -无线 接入网接口 (Iu)用户平面速率控制帧 49中的多个无线接入承载子流组合指示的序号标 识(RFCI n indicator) 域, 指定了一一对应的多个无线接入承载子流组合指示(RFCI), 媒体网关 (MGW) 的速率控制单元根据这些^ 5线接入承载子流组合指示 (RFCI-RAB sub-Flow Combination Indicator)产生指示多个自适应多速率(AMR)话音编码模式的模 式选择信号 40, 输出模式选择信号 40到码型变换器 (TC), 指定码型变换器 (TC) 输 出的话音编码帧序列 42中应包括的多个自适应多速率 (AMR) 编码帧的模式。 (MGW) output transmission core network - radio access network interface (Iu) user plane rate control frame 49, core network - radio access network interface (Iu) user plane rate control frame 49 multiple radio access bearer substreams a sequence identifier (RFCI n indicator) field of the combined indication, which specifies a plurality of radio access bearer substream combination indications (RFCIs) corresponding to one-to-one correspondence, The rate control unit of the media gateway (MGW) generates a mode selection signal 40 indicating a plurality of adaptive multi-rate (AMR) voice coding modes according to the RFCI-RAB sub-Flow Combination Indicator (RFCI-RAB sub-Flow Combination Indicator). The mode select signal 40 is output to a pattern converter (TC) specifying a pattern of a plurality of adaptive multi-rate (AMR) coded frames that should be included in the sequence of speech encoded frames 42 output by the pattern converter (TC).
核心网-无线接入网接口 (Iu)使用哪一种 线接入承载子流组合是由无线接入承载 子流组合指示(RFCI)给出, 在通信阶段无线接入承载子流组合指示(RFCI)被包含在 Iu用户平面帧 43中, 无线接入承载子流组合†旨示 (RFCI) 给出了核心网-无线接入网接 口 (Iu)用户平面帧 43的结构。  Which type of line access bearer substream combination is used by the core network-radio access network interface (Iu) is given by the radio access bearer substream combination indication (RFCI), and the radio access bearer substream combination indication is in the communication phase ( The RFCI) is included in the Iu User Plane Frame 43, and the Radio Access Bearer Substream Combination (RFCI) gives the structure of the Core Network-Radio Access Network Interface (Iu) User Plane Frame 43.
表格 9是无线网络控制器(RNC) 传送话音帧编码序列到核心网所使用的无线接入 承载子流组合(RFC)及其无线接入承载子流组 指示(RFCI)值的示例,它们也是 Iu UP 初始化时在核心网 -无线接入网接口用户平面 (Iu UP) 初始化过程中需要被发送的部分。  Table 9 is an example of a Radio Access Bearer Substream Combination (RFC) used by a Radio Network Controller (RNC) to transmit a voice frame coding sequence to a core network and its Radio Access Bearer Subflow Group Indicator (RFCI) values, which are also The part of the core network-radio access network interface user plane (Iu UP) initialization process that needs to be sent during Iu UP initialization.
在表格 10的示例中给出了: 利用核心网- 线接入网接口(Iu)用户平面速率控制帧 49的无线接入承载子流组合指示的序号标识 (RFCI n indicator) 向核心网中的媒体网关 (MGW)发出和 RFC对应关系的指示, 速率控制部分按照该指示确定多个自适应多速 率(AMR)编码模式, 输出含有所述多个自适应多速率(AMR)编码模式的模式选择信 号 40到码型变换器 (TC), 码型变换器 (TC) 根据收到的话音编码信号 41的类型作码 型转换, 核心网-无线接入网接口 (Iu) 功能单元选择适当的无线接入承载子流组合指示 (RFCI)来承载话音帧编码序列 42中的帧。  In the example of Table 10, the numbering identifier (RFCI n indicator) of the radio access bearer substream combination indication using the core network-line access network interface (Iu) user plane rate control frame 49 is given to the core network. The media gateway (MGW) issues an indication corresponding to the RFC, and the rate control portion determines a plurality of adaptive multi-rate (AMR) coding modes according to the indication, and outputs a mode selection including the plurality of adaptive multi-rate (AMR) coding modes. Signal 40 to the code converter (TC), the pattern converter (TC) performs pattern conversion according to the type of the received voice coded signal 41, and the core network-radio access network interface (Iu) functional unit selects the appropriate wireless The Access Bearer Substream Combination Indicator (RFCI) carries the frames in the voice frame encoding sequence 42.
表格 9  Form 9
RFCI 无 线 无 线 无 线 无 线 无 线 无 线 无 线 无 线RFCI Wireless, Wireless, Wireless, Wireless, Wireless, Wireless, Wireless
(无线接入 接 入 接 入 接 入 接 入 接 入 接 入 接 入 接 入 入承载 (wireless access, access, access, access, access, access, and bearer)
承载子流 承 载 承 载 承 载 承 载 承 载 承 载 承 载 承 载 子流 1  Bearing substream bearing load bearing load bearing load carrier load substream 1
组合指示) 子流 2 子流 3 子流 4 子流 5 子流 6 子流 7 子流 8 子流 9 Combination indication) Substream 2 Substream 3 Substream 4 Substream 5 Substream 6 Substream 7 Substream 8 Substream 9
0 0 0 0 0 0 0 0 0 00 0 0 0 0 0 0 0 0 0
1 39 0 0 0 0 0 0 0 01 39 0 0 0 0 0 0 0 0
2 42 53 0 0 0 0 0 0 02 42 53 0 0 0 0 0 0 0
3 61 87 0 0 0 0 0 0 03 61 87 0 0 0 0 0 0 0
4 81 103 60 0 0 0 0 0 04 81 103 60 0 0 0 0 0 0
5 42 53 0 61 87 0 0 0 05 42 53 0 61 87 0 0 0 0
6 42 53 0 81 103 60 0 0 06 42 53 0 81 103 60 0 0 0
7 61 87 0 81 103 60 0 0 07 61 87 0 81 103 60 0 0 0
8 42 53 0 61 87 0 81 103 60 表格 10 8 42 53 0 61 87 0 81 103 60 Form 10
Figure imgf000033_0001
实施例 5——网络侧以间断的方式传送自适应多速率(AMR)编码帧
Figure imgf000033_0001
Embodiment 5 - The network side transmits an adaptive multi-rate (AMR) coded frame in an intermittent manner
参见图 18,图 18的方案是图 17的改进,它和图 17的区别在于多出了先进先出存储 器(FIFO)和用传输格式组合选择先进先出存储器读取控制部件替代了图 17中的传输格 式组合选择部件。 码型变换器输出的话音帧编码序列 42像图 17中的一样, 经过 Iu接口 功能单元处理后先形成 Iu用户平面帧 43输入到无线网络控制器(RNC),再经 Iu接口功 能单元处理后还原成话音帧编码序列 42, 然后输出到到先进先出存储器(FIFO), 先进 先出存储器(FIFO)缓存话音帧编码序列, 并输出存储状态标志 421, 存储状态标志 421 指示: 是否有未被读取的话音帧编码序列, 以及这些未被读取的话音帧编码序列的数目。 传输格式组合选择先进先出存储器读取控制部件输出读取命令 422,读取命令 422使先进 先出存储器(FIFO) 输出话音帧编码序列 420。 Referring to FIG. 18, the scheme of FIG. 18 is a modification of FIG. 17, which differs from FIG. 17 in that a FIFO is added and a FIFO read control component is selected in combination with a transport format instead of FIG. The transport format combination selects the components. The voice frame coding sequence 42 outputted by the pattern converter is processed as shown in FIG. 17, and is processed by the Iu interface function unit to form an Iu user plane frame 43 and input to the radio network controller (RNC), and then processed by the Iu interface function unit. Reverting to the voice frame coding sequence 42, and then outputting to the first in first out memory (FIFO), the first in first out memory (FIFO) buffer voice frame coding sequence, and outputting the storage status flag 421, the storage status flag 421 indicating: whether there is any The sequence of voice frame codes read, and the number of sequences of voice frames that are not read. The transport format combination selects the first in first out memory read control unit output read command 422, and the read command 422 makes the advanced The first-out memory (FIFO) outputs a voice frame coding sequence 420.
为把话音的延迟控制在目的移动台的缓存区下限值对应的时间附近, 在存储状态标 志 421给出存储的话音帧编码序列的数目超过目的移动台的缓存区下限值的指示之后, 传输格式组合选择先进先出存储器读取控制部件可以用读取命令 422选择话音帧编码序 列将其从先进先出存储器(FIFO) 中读出并加以丢弃。  In order to control the delay of the voice near the time corresponding to the lower limit value of the buffer area of the destination mobile station, after the storage status flag 421 gives an indication that the number of stored voice frame coding sequences exceeds the lower limit value of the buffer area of the destination mobile station, Transport Format Combination Selection The first in first out memory read control component can read and discard the voice frame encoding sequence from the first in first out memory (FIFO) using the read command 422.
表格 11表示实施例在工作过程中的一个示例性的传输信道的配置,给出各传输信道 的属性和参数, 特别是每一传输信道的传输格式标识(TFI)所对应的传输格式(TF)和 传输时间间隔 (ΤΉ)。  Table 11 shows the configuration of an exemplary transport channel in the working process of the embodiment, giving the attributes and parameters of each transport channel, in particular the transport format (TF) corresponding to the transport format identifier (TFI) of each transport channel. And transmission time interval (ΤΉ).
表格 11 Form 11
Figure imgf000034_0001
用表格 9作为无线接入承载子流组合 (RFC)及其无线接入承载子流组合指示 (RFCI) 值的示例, 当核心网-无线接入网接口(Iu)用户平面速率控制帧 49的无线接入承载子流 组合指示的序号标识 (RFCI n indicator) 为 (0,1,2,4,6), 使得速率控制单元输出的模式 选择信号 40包含的模式为(12.2kbps, 4.75bps), 核心网在核心网-无线接入网接口 (Iu) 用户平面帧中使用值为 6的 RFCI来传输话音帧编码序列 42, 无线网络控制器(RNC) 存储在先进先出存储器(FIFO) 中的话音帧编码序列包含的是 12.2kbps和 4.75bps模式 的话音数据, 如果在需要调度到传输信道信令数据 5中有有效数据需要传送, 分组数据 中没有有效数据, 且由于物理信道带宽的限制传送格式组合(2, 1 , 1, 0, 1 )是无效组 合, 这样无线接口功能单元用传送格式组合(0, 0, 0, 0, 1 )在一个 20毫秒的传输时 间间隔内发送完信令数据后, 无线接口功能单元可以用 (4, 3, 0, 0, 0)在连续 2个 10 毫秒的传输时间间隔发送模式为 4.75kbps的话音帧, 使得, 受信令数据抢先而暂停发送 的话音数据得到了延迟的发送, 同时又不影响后续的话音帧的发送。 当目标移动台的缓存区长度为 4个帧时, 在发生切换时无线网络控制器无法发送无 线帧, 当切换期间无线网络控制器从核心网处总共收到了 8个话音帧编码序列, 无线网 络控制器确定以 5 (4+1 )作为超时参数, 当存储状态标志 421给出存储的话音帧编码序 列的数目大于 5 时丢弃超时的序列, 这些话音帧编码序列同上一个示例一样, 是 ( 12.2kbps, 4.75bps) 的模式, 切换完成后无线接口功能单元可以用 (4, 3, 0, 0, 0) 在连续 5个 10毫秒的传输时间间隔发送切换期间的话音编码帧序列中的模式为 4.75kbps 的帧, 接下来再用若干个 10毫秒的传输时间间隔发送完先进先出存储器(FIFO)内存放 时间超过 20毫秒的延迟话音帧编码序列,然后就可以恢复模式为 12.2kbps的话音帧发送 了。
Figure imgf000034_0001
Table 9 is used as an example of a Radio Access Bearer Substream Combination (RFC) and its Radio Access Bearer Substream Combination Indicator (RFCI) value, when the Core Network-Radio Access Network Interface (Iu) User Plane Rate Control Frame 49 The sequence identifier (RFCI n indicator) indicated by the radio access bearer substream combination is (0, 1, 2, 4, 6), so that the mode selection signal 40 output by the rate control unit includes a mode (12.2 kbps, 4.75 bps). The core network transmits the voice frame coding sequence 42 using the RFCI with a value of 6 in the core network-radio access network interface (Iu) user plane frame, and the radio network controller (RNC) is stored in the first in first out memory (FIFO). The voice frame coding sequence contains voice data in the 12.2 kbps and 4.75 bps modes. If there is valid data to be transmitted in the transmission channel signaling data 5, there is no valid data in the packet data, and due to the limitation of the physical channel bandwidth. The transport format combination (2, 1, 1, 0, 1) is an invalid combination, so that the radio interface function unit sends the message with a transport format combination (0, 0, 0, 0, 1) within a 20 ms transmission time interval. After the data, the wireless interface function unit can use (4, 3, 0, 0, 0) to transmit a voice frame of 4.75 kbps in consecutive 2 10 millisecond transmission time intervals, so that the voice is suspended due to the preemption of the signaling data. The data is delayed in transmission without affecting the subsequent transmission of voice frames. When the buffer length of the target mobile station is 4 frames, the radio network controller cannot transmit the radio frame when the handover occurs, and the radio network controller receives a total of 8 voice frame coding sequences from the core network during the handover, the wireless network The controller determines 5 (4+1) as the timeout parameter, and discards the timeout sequence when the stored status flag 421 gives the number of stored voice frame coding sequences greater than 5, and these voice frame coding sequences are the same as the previous example, ( 12.2 Kbps, 4.75bps) mode, after the switch is completed, the wireless interface function unit can use (4, 3, 0, 0, 0) to transmit the pattern in the sequence of voice coded frames during the switching interval of 5 consecutive 10 milliseconds. 4.75 kbps frame, then send a delayed voice frame coding sequence with a storage time of more than 20 milliseconds in the first-in first-out memory (FIFO) with several 10 millisecond transmission time intervals, and then recover the voice frame with the mode of 12.2 kbps. Sent.
实施例 6——网络侧传送被延迟的自适应多速率 (AMR)编码帧  Embodiment 6 - Network side transmission delayed adaptive multi-rate (AMR) coded frame
图 18指示的实施例 5也可以用于核心网向无线网络控制器 (R C)发送带延迟标志 的话音编码帧序列的处理, 延迟标志可以放在核心网-无线接入网接口 (Iu)用户平面帧 的扩展域(spare extension)里, 带延迟标志的话音编码帧序列是由发送端移动台发送的 被延迟的自适应多速率话音编码帧引起的, 例如, 发送端移动台由于切换在发送的缓存 区内积累了多个话音帧编码序列, 导致切换完成后要发送这些被延迟的话音帧编码序列 中的自适应多速率话音编码帧。带延迟标志的话音编码帧序列总是在 20毫秒内和其他编 码帧序列一起到达无线网络控制器 (RNC), 所以, 无线网络控制器(R C) 不在 20毫 秒的时间内发送单独一个带延迟标志的 AMR编码帧, 以在每次 20毫秒的发送时间发送 多个帧的方式来在传输信道上调度这些带延迟标志的话音编码帧。  The embodiment 5 indicated in FIG. 18 can also be used for processing the core network to transmit a sequence of voice coded frames with delay flags to the radio network controller (RC), and the delay flag can be placed on the core network-radio access network interface (Iu) user. In the spare extension of the plane frame, the sequence of voice coded frames with delay flags is caused by the delayed adaptive multi-rate voice coded frame transmitted by the mobile station at the transmitting end. For example, the mobile station at the transmitting end is transmitting due to handover. A plurality of voice frame coding sequences are accumulated in the buffer area, so that the adaptive multi-rate voice coded frames in the delayed voice frame coding sequences are transmitted after the handover is completed. A sequence of voice coded frames with a delay flag always arrives at the Radio Network Controller (RNC) with other coded frame sequences within 20 milliseconds, so the Radio Network Controller (RC) does not send a single delay flag within 20 milliseconds. The AMR encoded frame schedules the speech encoded frames with delay flags on the transport channel in a manner that multiple frames are transmitted at a transmission time of 20 milliseconds each.
表格 12表示实施例在工作过程中的一个示例性的传输信道的配置,给出各传输信道 的属性和参数, 特别是每一传输信道的传输格式标识(TFI)所对应的传输格式(TF)和 传输时间间隔 (ΤΉ)。  Table 12 shows the configuration of an exemplary transport channel in the working process of the embodiment, giving the attributes and parameters of each transport channel, in particular the transport format (TF) corresponding to the transport format identifier (TFI) of each transport channel. And transmission time interval (ΤΉ).
表格 13是实施例在工作过程中所作的调度话音编码帧到传输信道的示例。  Table 13 is an example of scheduling a voice coded frame to a transmission channel made by an embodiment during operation.
表格 12 传输信道 1 2 3  Table 12 Transport Channels 1 2 3
信道编码 Turbo 卷积 卷积  Channel coding Turbo convolution convolution
编码速率 1/3 1/2 1/2  Coding rate 1/3 1/2 1/2
0 1 X 0, 20ms 0, 20ms 0, 20ms  0 1 X 0, 20ms 0, 20ms 0, 20ms
1 1 X 39, 20ms I X 103, 20ms 1 X 60, 20ms  1 1 X 39, 20ms I X 103, 20ms 1 X 60, 20ms
TFI 2 1 X 81, 20ms 1 X 87, 20ms  TFI 2 1 X 81, 20ms 1 X 87, 20ms
3 1 X 61 , 20ms 1 X 53, 10ms  3 1 X 61 , 20ms 1 X 53, 10ms
4 1 X42, 10ms 表格 13 先进先出存储器中的话 所使用的 TFI 注释 4 1 X42, 10ms Table 13 TFI notes used in FIFO memory
音编码帧序列 Tone coded frame sequence
(4.75kbps, 延迟标志) (4, 3, 0)  (4.75kbps, delay flag) (4, 3, 0)
(4.75kbps) (4, 3, 0)  (4.75kbps) (4, 3, 0)
(4.75kbp,延迟标志) (4, 3, 0) 在表格 12 中无法找到 10 毫秒的(4.7 5 kbp, delay flag) (4, 3, 0) Cannot find 10 ms in Table 12
(4;75kbps,7.4kbps) (4, 3, 0) 7.4kbps 模式的传输格式, 只能不用 (4; 75 kbps, 7.4 kbps) (4, 3, 0) 7.4 kbps mode of transmission format, can only be used
7.4kbps模式而改用 4.75kbps模式 7.4kbps mode instead of 4.75kbps mode
(4.75kbps, 延迟标志) (4, 3, 0) (4.75kbps, delay flag) (4, 3, 0)
(4.75kbps, 延迟标志) (4, 3, 0)  (4.75kbps, delay flag) (4, 3, 0)

Claims

权利要求 Rights request
1、 一种在移动台中生成和传送自适应多速率 (AMR) 编码帧的方法, 在话音呼叫 过程中, 自适应多速率 (AMR) 编码器通过对输入话音信号的采样得到长度为 20毫秒 的话音帧, 每个话音帧含 20毫秒话音的采样样本, 对 20毫秒话音帧编码, 所述编码产 生的是符合通用移动通信***(UMTS)标准的自适应多速率(AMR) 编码帧, 需要无 线接口功能单元发送自适应多速率 (AMR)编码帧形式的话音数据, 无线接口功能单元 按各传输信道的传输格式发送所述的话音数据, 其特征在于: 1. A method of generating and transmitting an adaptive multi-rate (AMR) coded frame in a mobile station, the adaptive multi-rate (AMR) encoder obtaining a length of 20 milliseconds by sampling the input voice signal during a voice call Voice frames, each voice frame containing 20 millisecond voice sample samples, encoding a 20 millisecond voice frame, which produces an adaptive multi-rate (AMR) coded frame conforming to the Universal Mobile Telecommunications System (UMTS) standard, requiring wireless The interface function unit transmits voice data in the form of an adaptive multi-rate (AMR) coded frame, and the wireless interface function unit transmits the voice data according to a transmission format of each transport channel, which is characterized by:
向自适应多速率(AMR) 编码器发出编码命令, 所述编码命令指定了多个自适应多 速率 (AMR) 模式, 自适应多速率 (AMR)编码器根据所述编码命令对一个 20毫秒话 音帧编码所产生的有效话音帧编码序列, 要么是多个自适应多速率 (AMR) 话音编码帧 的集合, 要么是一个自适应多速率 (AMR)静默编码帧, 当所述话音帧编码序列是多个 自适应多速率(AMR)话音编码帧时, 其自适应多速率(AMR)编码帧的模式同所述编 码命令中的自适应多速率 (AMR) 模式相一致;  An encoding command is issued to an adaptive multi-rate (AMR) encoder, the encoding command specifying a plurality of adaptive multi-rate (AMR) modes, the adaptive multi-rate (AMR) encoder according to the encoding command to a 20 millisecond voice The effective voice frame coding sequence generated by frame coding is either a set of multiple adaptive multi-rate (AMR) voice coded frames, or an adaptive multi-rate (AMR) silence coded frame, when the voice frame coding sequence is When multiple adaptive multi-rate (AMR) speech encoded frames, the mode of the adaptive multi-rate (AMR) encoded frame is consistent with the adaptive multi-rate (AMR) mode in the encoding command;
向无线接口功能单元输出话音帧编码序列;  Outputting a voice frame coding sequence to the wireless interface function unit;
在无线接口功能单元中, 发送一个话音帧编码序列中的自适应多速率(AMR) 编码 帧到传输信道的过程, 包括, 在该话音帧编码序列中挑选一个自适应多速率(AMR)编 码帧和挑选一个传输格式组合, 挑选出的传输格式组合, 包含了传送挑选出的自适应多 速率(AMR) 编码帧所有类比特的传输格式, 用该传输格式组合将挑选出的自适应多速 率 (AMR) 编码帧调度到传输信道上。  In the wireless interface functional unit, the process of transmitting an adaptive multi-rate (AMR) coded frame in a voice frame coding sequence to a transmission channel, comprising selecting an adaptive multi-rate (AMR) coded frame in the voice frame coding sequence And selecting a transport format combination, the selected transport format combination, including the transmission format of all the bits of the selected adaptive multi-rate (AMR) coded frame, and using the transport format combination to select the adaptive multi-rate ( AMR) The coded frame is scheduled onto the transport channel.
2、 按照权利要求 1的方法, 其特征在于:  2. A method according to claim 1, characterized in that:
所述编码命令中的每个自适应多速率(AMR)模式同移动台的传输格式组合集中的 一个话音传输格式组合相对应,所述话音传输格式组合是包含一种自适应多速率(AMR) 模式编码帧所有类比特传输格式的传输格式组合。  Each of the adaptive multi-rate (AMR) modes of the encoding command corresponds to a combination of voice transmission formats in a combination of transmission formats of the mobile station, the voice transmission format combination comprising an adaptive multi-rate (AMR) Mode Coding Frame A combination of transport formats for all class bit transmission formats.
3、 按照权利要求 2的方法, 其特征在于, 在移动台暂停发送承载话音数据的无线帧 期间对来自自适应多速率 (AMR) 编码器的话音帧编码序列进行缓存, 在所述暂停结束 后, 在传送话音帧编码序列中的自适应多速率 (AMR) 编码帧到传输信道的过程中, 以 每 20毫秒调度一个以上自适应多速率(AMR)编码帧到传输信道的方式, 处理被缓存 的部分或全部话音帧编码序列。  3. A method according to claim 2, wherein the sequence of voice frame codes from the adaptive multi-rate (AMR) encoder is buffered during the suspension of the transmission of the radio frame carrying the voice data, after the suspension is completed During the process of transmitting an adaptive multi-rate (AMR) coded frame in the voice frame coding sequence to the transmission channel, scheduling more than one adaptive multi-rate (AMR) coded frame to the transmission channel every 20 milliseconds, the process is buffered Part or all of the voice frame coding sequence.
4、按照权利要求 3的方法, 其特征在于, 所述移动台暂停发送无线帧期间指小区切 换期间。  The method according to claim 3, characterized in that said mobile station pauses to transmit a radio frame period during a cell switching period.
5、按照权利要求 3的方法, 其特征在于, 所述移动台暂停发送无线帧期间指在优先 调度高于话音逻辑信道优先级的其它逻辑信道到传送信道时的暂停话音数据发送期间。  The method according to claim 3, characterized in that said mobile station suspending transmission of the radio frame period refers to a period of pause voice data transmission when preferentially scheduling other logical channels higher than the voice logical channel priority to the transmission channel.
6、 按照权利要求 3至 5中任意一项的方法, 其特征在于, 所述每 20毫秒调度一个 以上 (不包括一个) 自适应多速率 (AMR)编码帧到传输信道的方式是, 调度一个完整 的自适应多速率(AMR) 编码帧到 10毫秒传输时间间隔 (TTI) 的传输信道。 6. A method according to any one of claims 3 to 5, characterized in that said one is scheduled every 20 milliseconds The above (excluding one) adaptive multi-rate (AMR) coded frame to transport channel is to schedule a complete adaptive multi-rate (AMR) coded frame to a 10 millisecond transmission time interval (TTI) transport channel.
7、 一种移动台的处理接收到的自适应多速率 (AMR) 编码帧的方法, 移动台从传 送话音的传输信道上接收自适应多速率(AMR)编码帧, 自适应多速率(AMR)译码器 对移动台收到的自适应多速率 (AMR)编码帧按从先到后的次序每 20毫秒完成一个自 适应多速率 (AMR) 编码帧的译码输出, 其特征在于:  7. A method of processing a received adaptive multi-rate (AMR) coded frame by a mobile station, the mobile station receiving an adaptive multi-rate (AMR) coded frame from a transport channel for transmitting voice, an adaptive multi-rate (AMR) The decoder performs an adaptive multi-rate (AMR) coded frame decoding output every 20 milliseconds in an adaptive multi-rate (AMR) coded frame received by the mobile station in a first to last order, characterized by:
为一个话音呼叫设立放置自适应多速率(AMR) 编码帧的一个缓存区和该缓存区的 下限值, 从移动台第一次收到所述话音呼叫的自适应多速率(AMR) 编码帧开始, 将收 到的自适应多速率(AMR)话音编码帧存放到缓存区内,当缓存区内自适应多速率 (AMR) 编码帧的个数达到所述下限值时, 自适应多速率 (AMR)译码器开始对移动台收到的自 适应多速率 (AMR) 编码帧译码,  Establishing a buffer area for placing an adaptive multi-rate (AMR) coded frame and a lower limit value of the buffer for a voice call, and receiving an adaptive multi-rate (AMR) coded frame of the voice call from the mobile station for the first time Initially, the received adaptive multi-rate (AMR) voice coded frame is stored in the buffer area, and when the number of adaptive multi-rate (AMR) coded frames in the buffer area reaches the lower limit value, the adaptive multi-rate The (AMR) decoder begins decoding the adaptive multi-rate (AMR) coded frame received by the mobile station.
当移动台处于语音(SPEECH)模式时,检査每 20毫秒缓存区内自适应多速率(AMR) 话音编码帧增加的的数目: 当所述数目小于 1时, 主动添加自适应多速率 (AMR) 接收 丢失帧; 当所述数目大于 1并且缓冲区中存在语音(SPEECH)模式下主动添加的接收丢 失帧时, 删除所述的在语音 (SPEECH)模式下主动添加的接收丢失帧。  When the mobile station is in voice (SPEECH) mode, check the number of adaptive multi-rate (AMR) voice coded frames added in the buffer area every 20 milliseconds: When the number is less than 1, actively add adaptive multi-rate (AMR) Receiving a lost frame; when the number is greater than 1 and the received lost frame is actively added in the voice (SPEECH) mode in the buffer, the received lost frame actively added in the voice (SPEECH) mode is deleted.
8、 按照权利要求 7的方法, 其特征在于- 设立所述缓存区的在语音 (SPEECH)模式下主动添加的接收丢失帧的个数限制值, 当缓存区内语音(SPEECH)模式下主动添加的接收丢失帧的个数达到所述的个数限制值 时, 在语音 (SPEECH)模式下不再添加接收丢失帧。  8. The method according to claim 7, characterized in that - the number of received lost frames that are actively added in the voice (SPEECH) mode of the buffer area is set up, and is actively added when the voice in the buffer area (SPEECH) mode is added. When the number of received lost frames reaches the number limit, the received lost frame is no longer added in the voice (SPEECH) mode.
9、按照权利要求 7至 8中任一项的方法,其特征在于,移动台向网络发送含所述缓存 区的下限值的消息。  A method according to any one of claims 7 to 8, characterized in that the mobile station transmits a message containing the lower limit value of said buffer area to the network.
10、 一种生成和传送自适应多速率(AMR)编码帧的方法, 在话音呼叫过程中, 在 核心网中的码型变换器(TC)将输入的话音编码信号转换成自适应多速率(AMR)编码 帧, 所述自适应多速率(AMR)编码帧被放置到核心网-无线接入网接口 (Iu)用户平面 帧上向无线网络控制器(RNC)发送, 无线网络控制器(RNC)将自适应多速率(AMR) 编码帧调度到传输信道上, 其特征在于:  10. A method of generating and transmitting an adaptive multi-rate (AMR) coded frame, wherein a code converter (TC) in a core network converts an input voice coded signal into an adaptive multi-rate during a voice call ( AMR) coded frame, the adaptive multi-rate (AMR) coded frame is placed on a core network-radio access network interface (Iu) user plane frame and sent to a radio network controller (RNC), the radio network controller (RNC) Scheduling adaptive multi-rate (AMR) encoded frames onto the transport channel, characterized by:
向码型变换器发送编码命令,所述编码命令指定了多个自适应多速率(AMR)模式, 码型变换器按所述编码命令, 对输入的话音编码信号进行编码操作, 所产生的有效的 20 毫秒长度话音的话音帧编码序列, 要么是若干 (大于等于 1 )个自适应多速率 (AMR) 话音编码帧的集合, 要么是一个自适应多速率 (AMR) 静默编码帧, 要么是一个无数据 (NO— DATA)类型帧, 当所述话音帧编码序列是多 (大于 1 )个自适应多速率(AMR) 话音编码帧时, 其自适应多速率 (AMR)编码帧的模式同所述编码命令中的自适应多速 率 (AMR)模式相一致;  Transmitting an encoding command to the pattern converter, the encoding command specifying a plurality of adaptive multi-rate (AMR) modes, and the pattern converter performs an encoding operation on the input speech encoded signal according to the encoding command, and the generated encoding is effective The 20 ms long speech voice frame coding sequence, either a set of several (greater than or equal to 1) adaptive multi-rate (AMR) speech coded frames, either an adaptive multi-rate (AMR) silence coded frame, or a No data (NO-DATA) type frame, when the voice frame coding sequence is multiple (greater than 1) adaptive multi-rate (AMR) voice coded frames, the mode of the adaptive multi-rate (AMR) coded frame is the same The adaptive multi-rate (AMR) mode in the coding command is consistent;
用核心网-无线接入网接口 (Iu)用户平面帧承载话音帧编码序列中的所有自适应多 速率 (AMR)编码帧, 向无线网络控制器(RNC) 发送所述承载话音帧编码序列的核心 网-无线接入网接口 (Iu) 用户平面帧; Using the core network-radio access network interface (Iu) user plane frame to carry all the adaptive multiples in the voice frame coding sequence Rate (AMR) encoded frame, transmitting the core network-radio access network interface (Iu) user plane frame carrying the voice frame coding sequence to a radio network controller (RNC);
在无线网络控制器(RNC)中,调度一个话音帧编码序列中的自适应多速率(AMR) 编码帧到传输信道的过程,包括,在该话音帧编码序列中挑选一个自适应多速率(AMR) 编码帧和挑选一个传输格式组合, 挑选出的传输格式组合, 包含了传送挑选出的自适应 多速率 (AMR) 编码帧所有类比特的传输格式, 用该传输格式组合将挑选出的自适应多 速率 (AMR)编码帧调度到传输信道上。  In a Radio Network Controller (RNC), the process of scheduling an adaptive multi-rate (AMR) coded frame in a voice frame coding sequence to a transmission channel, including selecting an adaptive multi-rate (AMR) in the sequence of voice frame codes Encoding the frame and selecting a combination of transmission formats, and selecting the selected combination of transmission formats, including the transmission format of all the bits of the selected adaptive multi-rate (AMR) coded frame, and using the combination of the transmission formats to select the adaptive Multi-rate (AMR) coded frames are scheduled onto the transport channel.
11、 按照权利要求 10的方法, 其特征在于:  11. A method according to claim 10, characterized in that:
无线网络控制器 (RNC) 向核心网发送核心网-无线接入网接口 (Iu)用户平面速率 控制帧, 核心网-无线接入网接口 (Iu)用户平面速率控制帧中的多个无线接入承载子流 组合指示的序号标识 (RFCI n indicator)域与多个无线接入承载子流组合指示(RFCI) 一一对应, 这些无线接入承载子流组合指示(RFCI)又与自适应多速率(AMR)模式相 对应, 核心网收到所述核心网-无线接入网接口 (Iu) 用户平面速率控制帧后, 向码型变 换器发送的编码命令, 编码命令中指定的多个自适应多速率 (AMR) 模式就是所述无线 接入承载子流组合指示 (RFCI)所对应的自适应多速率(AMR) 话音模式。  The Radio Network Controller (RNC) sends a Core Network-Radio Access Network Interface (Iu) User Plane Rate Control Frame to the Core Network, and Multiple Radio Connections in the Core Network-Radio Access Network Interface (Iu) User Plane Rate Control Frame The sequence number identifier (RFCI n indicator) field of the incoming bearer stream combination indication is in one-to-one correspondence with multiple radio access bearer substream combination indications (RFCIs), and the radio access bearer substream combination indication (RFCI) is more adaptive than Corresponding to the rate (AMR) mode, after the core network receives the core network-radio access network interface (Iu) user plane rate control frame, the coding command sent to the pattern converter, the multiple specified in the coding command The adaptive multi-rate (AMR) mode is an adaptive multi-rate (AMR) voice mode corresponding to the Radio Access Bearer Substream Combination Indicator (RFCI).
12、 按照权利要求 11的方法, 其特征在于, 无线网络控制器 (R C)设立缓存区, 将暂停无线帧发送期间收到的由核心网发送的 Iu用户平面帧所承载的话音帧编码序列缓 存在该缓存区, 所述暂停结束后, 在调度被缓存的话音帧编码序列中的自适应多速率 12. The method according to claim 11, characterized in that the radio network controller (RC) sets up a buffer area for buffering the voice frame coding sequence buffer carried by the Iu user plane frame transmitted by the core network received during the transmission of the radio frame. In the buffer area, after the pause is finished, the adaptive multi-rate in the sequence of the buffered voice frame coding is scheduled.
(AMR)编码帧到传输信道的过程中, 采用每 20毫秒调度一个以上 (不包括一个) 自 适应多速率(AMR) 编码帧到传输信道的方式。 (AMR) In the process of encoding a frame to a transmission channel, one or more (excluding one) adaptive multi-rate (AMR) coded frames are transmitted every 20 milliseconds to the transmission channel.
13、按照权利要求 12的方法, 其特征在于, 所述暂停无线帧发送期间是由无线网络 控制器 (RNC)之内的小区间切换期间。  13. A method according to claim 12, wherein said paused radio frame transmission period is during an inter-cell handover period within a Radio Network Controller (RNC).
14、按照权利要求 12的方法, 其特征在于, 所述暂停无线帧发送期间是无线网络控 制器 (RNC)间切换期间。  The method of claim 12 wherein said paused radio frame transmission period is during a radio network controller (RNC) handover.
15、按照权利要求 12的方法, 其特征在于, 所述暂停无线帧发送期间是由优先调度 髙于话音逻辑信道优先级的其它逻辑信道到传送信道时的暂停话音数据发送所引起的未 能发送那些核心网-无线接入网接口 (Iu)用户平面帧所承载的话音帧编码序列中自适应 多速率(AMR) 编码帧的期间。  15. The method according to claim 12, wherein said suspending radio frame transmission period is a failure to transmit due to suspending voice data transmission when preferentially scheduling other logical channels prior to the voice logical channel priority to the transmission channel The period of adaptive multi-rate (AMR) encoded frames in the voice frame coding sequence carried by the core network-radio access network interface (Iu) user plane frame.
16、按照权利要求 11的方法,其特征在于,所述核心网发送到无线网络控制器 (RNC) 的承载话音帧编码序列的 Iu用户平面帧包含该话音帧编码序列的延迟标志, 当延迟标志 指示话音帧编码序列是被延迟了的时, 无线网络控制器(R C) 以每 20毫秒调度一个以 上 (不包括一个) 自适应多速率 (AMR) 编码帧到传输信道的方式来调度该话音帧编码 序列。  The method according to claim 11, wherein the Iu user plane frame of the voice network frame coding sequence transmitted by the core network to the radio network controller (RNC) comprises a delay flag of the voice frame coding sequence, when the delay flag is When the voice frame coding sequence is delayed, the radio network controller (RC) schedules the voice frame by scheduling more than one (excluding one) adaptive multi-rate (AMR) coded frame to the transmission channel every 20 milliseconds. Coding sequence.
17、按照权利要求 12至 16中任一项的方法, 其特征在于, 所述每 20毫秒调度一个 以上 (不包括一个) 自适应多速率(AMR)编码帧到传输信道的方式是, 调度一个完整 的自适应多速率 (AMR) 编码帧到 10毫秒传输时间间隔 (TTI) 的传输信道。 Method according to any one of claims 12 to 16, characterized in that said one is scheduled every 20 milliseconds The above (excluding one) adaptive multi-rate (AMR) coded frame to transport channel is to schedule a complete adaptive multi-rate (AMR) coded frame to a 10 millisecond transmission time interval (TTI) transport channel.
18、 按照权利要求 12至 16中任一项的的方法, 其特征在于: 当缓存的话音帧编码 序列的数目超过目的移动台的缓存区下限值后,开始丢弃缓存的话音帧编码序列。  18. A method according to any one of claims 12 to 16, characterized in that, after the number of buffered speech frame coding sequences exceeds the lower limit of the buffer area of the destination mobile station, the buffered speech frame coding sequence is discarded.
19、 一种通用移动通信***中的移动台, 该移动台的自适应多速率 (AMR)编码器 对 20毫秒长度话音帧编码产生话音帧编码序列的输出,并且该移动台包括一与自适应多 速率(AMR)编码器输出端相连接的先进先出 (FIFO)缓存, 该先进先出 (FIFO)缓存 包括:  19. A mobile station in a universal mobile communication system, the mobile station's adaptive multi-rate (AMR) encoder encoding a 20 ms long speech frame encoding to produce an output of a speech frame encoding sequence, and the mobile station includes an adaptive A first-in, first-out (FIFO) buffer connected to the output of the multi-rate (AMR) encoder, the first-in-first-out (FIFO) cache includes:
一数据输入接口, 用于读入自适应多速率 (AMR)编码器编码产生的话音帧编码序 列,  a data input interface for reading a sequence of voice frame codes generated by an adaptive multi-rate (AMR) encoder encoding,
一数据输出接口, 用于从该先进先出 (FIFO) 缓存读取被存储的话音帧编码序列, 一存储状态接口, 用于输出该先进先出 (FIFO)缓存中存储的话音帧编码序列的数 量和其中的话音帧编码序列的类型;  a data output interface for reading a stored voice frame code sequence from the first in first out (FIFO) buffer, a memory state interface for outputting a voice frame code sequence stored in the first in first out (FIFO) buffer The number and type of voice frame coding sequence therein;
从所述先进先出(FIFO)缓存的存储状态接口读取待处理的话音帧编码序列的类型, 当待处理的话音帧编码序列是需要发送的有效话音帧编码序列时, 为该话音帧编码序列 选择传输格式组合, 用所选择的传输格式组合传输该话音帧编码序列中的一个自适应多 速率 (AMR)编码帧。  Reading the type of the voice frame coding sequence to be processed from the storage state interface of the first in first out (FIFO) buffer, and encoding the voice frame when the voice frame coding sequence to be processed is a valid voice frame coding sequence to be transmitted The sequence selects a transport format combination to transmit an adaptive multi-rate (AMR) coded frame in the voice frame coding sequence in combination with the selected transport format.
20、 如权利要求 19 的移动台, 其特征在于, 在所述先进先出 (FIFO) 缓存存储 2 个以上的话音帧编码序列。  20. The mobile station of claim 19, wherein more than two voice frame coding sequences are stored in said first in first out (FIFO) buffer.
21、 如权利要求 19或 20的移动台, 其特征在于, 为所述的需要发送的待处理的话 音帧编码序列选择传输时间间隔 (ΤΉ) 可变的传输格式组合。  A mobile station according to claim 19 or 20, wherein a transmission time interval (ΤΉ) variable transmission format combination is selected for said to-be-processed voice frame coding sequence to be transmitted.
22、如权利要求 21的移动台, 其特征在于, 当所述先进先出 (FIFO)中存储的话音 帧编码序列的数量超过指定的数目时, 优选传输时间间隔 (TTI) 为 10毫秒的传输格式 组合。  22. The mobile station of claim 21, wherein when the number of voice frame coding sequences stored in said first in first out (FIFO) exceeds a specified number, preferably a transmission time interval (TTI) of 10 milliseconds is transmitted. Format combination.
' 23、 如权利要求 22的移动台, 其特征在于, 所述指定的数目为 1。  A mobile station according to claim 22, wherein said specified number is one.
24、 一种通用移动通信***中的移动台, 该移动台包括一与自适应多速率(AMR) 译码器输入端相连接的先进先出 (FIFO) 缓存, 该先进先出 (FIFO) 缓存包括:  24. A mobile station in a universal mobile communication system, the mobile station including a first in first out (FIFO) buffer coupled to an input of an adaptive multi-rate (AMR) decoder, the first in first out (FIFO) buffer Includes:
一数据输入接口, 用于读入自适应多速率 (AMR)编码帧,  a data input interface for reading in an adaptive multi-rate (AMR) encoded frame,
一数据输出接口,用于从该先进先出(FIFO)缓存读出被存储的自适应多速率 (AMR) 编码帧,  a data output interface for reading the stored adaptive multi-rate (AMR) encoded frame from the first in first out (FIFO) buffer,
一存储状态接口,用于输出该先进先出(FIFO)缓存中存储的自适应多速率(AMR) 编码帧队列的长度和编码帧的类型, 一控制单元, 用于根据删除指令删除存储在先进先出 (FIFO) 缓存内的自适应多速 率(AMR)编码帧队列中的无数据(NO— DATA)类型的自适应多速率(AMR)编码帧。 a storage status interface for outputting the length of the adaptive multi-rate (AMR) encoded frame queue stored in the first in first out (FIFO) buffer and the type of the encoded frame, a control unit, configured to delete the no-data (NO-DATA) type of adaptive multi-rate (AMR) coding in the adaptive multi-rate (AMR) coded frame queue stored in the first-in-first-out (FIFO) buffer according to the delete instruction frame.
25、 如权利要求 24 的移动台, 其特征在于, 在所述先进先出 (FIFO)缓存存储 2 个以上的自适应多速率(AMR)编码帧后开始向自适应多速率(AMR)译码器输入自适 应多速率 (AMR) 编码帧。  25. The mobile station of claim 24, wherein the adaptive multi-rate (AMR) decoding begins after the first in first out (FIFO) buffer stores more than two adaptive multi-rate (AMR) coded frames. Enter an adaptive multi-rate (AMR) encoded frame.
PCT/CN2005/001803 2004-11-11 2005-10-31 Method and device for adaptive multi-rate coding and transporting speech WO2006050657A1 (en)

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