Technical Field
[0001] The present invention relates to voice compensation technology, and more specifically,
to a method and system for voice compensation when the device at network side does
not use or partially use a vocoder.
Background
[0002] In a mobile communication system, vocoder at network side mainly has the following
two important roles: in the uplink, after the user equipment (UE) sends the compressively
encoded voice to the network side, the vocoder at the network side decodes the received
compressed voice to make the voice suitable to be transmitted in the network; in the
downlink, the vocoder at the network side compressively encodes the voice stream transmitted
in the network to make the voice suitable to be transmitted in the radio link.
[0003] Taking CDMA2000 (Code Division Multiple Access 2000) as an example, there are three
methods for voice encoding and decoding in CDMA2000: EVRC (Enhanced Variable Rate
Coder), QCELP-13k (Qualcomm Code Excited Linear Predictive Coding-Qualcomm-13k) and
QCELP-8k (Qualcomm Code Excited Linear Predictive Coding-Qualcomm-8k). Wherein, EVRC
is the most-widely used mainstream encoding and decoding format. In a typical call
between MS1 (Mobile Station) and MS2, they use the same voice encoding and decoding
method (such as EVRC). The voice of the MS1 subscriber arrives at the ears of the
MS2 subscriber through the following way: Firstly, MS1 sends the encoded EVRC compressed
voice frame to network side 1 through the radio uplink, and network side 1 uses vocoder
1 to decode the received EVRC voice frames and transform them into circuit PCM (Pulse
Coded Modulation) code stream, and then perform circuit switching. After network side
2 receives the circuit-switched PCM code stream from network side 1, it uses vocoder
2 to transform the PCM code stream to EVRC compressed voice frames and sends them
to MS2 through radio downlink.
[0004] The voice encoding and decoding by the vocoder is lossy compression, each encoding
and decoding will decrease the voice quality. Still taking the call between MS1 and
MS2 as an example, since MS1 and MS2 use the same encoding and decoding format, two
voice encoding and decoding processes can be avoided at the network side if the encoding
and decoding of EVRC compressed voice frames added at the network side can be avoided,
and the process that the voice of MS1 subscriber arrives at the ears of MS2 subscriber
is as follows: firstly, MS1 sends the encoded EVRC compressed voice frames to network
side 1 through radio uplink, and network side 1 directly switches the received EVRC
voice frames and sends them to network side 2; network side 2 receives the switched
EVRC compressed voice frames and sends them to MS2 through radio downlink.
[0005] From this example, it can be seen that, since two voice encoding and decoding processes
which will damage the voice quality are avoided at the network side, not only the
voice quality is evidently improved, but also the vocoder resource at the network
side is saved, thus reducing the delay for transmitting and processing the voice.
During the initial development stage of mobile communication system, the voice calls
in system are mainly between mobile subscribers and fixed subscribers, therefore,
the above influence is not evident. Traffic statistics shows that, the calls among
the mobile subscribers have already taken the leading position, and thus the original
vocoder configuration will not only add the device cost but also affect the system
performance. Therefore, how to improve the network structure and strategy of the vocoder
configuration management becomes the research hotspot.
[0006] Patent application
WO 02/054744 A1 discloses a network element and a method for enhancing the quality of digitised analogue
signals transmitted in parameterised coded form via a digital network. According to
this patent application, in order to enable an enhancement of the quality of the digitised
analogue signals on network side, the network element comprises means for extracting
signals from and insert signals into the network, first processing means for processing
the extracted parameters in the parameter domain with functions suitable to enhance
the quality of the digitised analogue signals and second processing means for processing
the extracted parameters in the linear domain with functions suitable to enhance the
quality of the digitised analogue signals.
[0007] With the development of all-IP mobile communication technology, supporting the traditional
voice service and packet data service with lower cost and more flexible method is
the main driving force of the development of all-IP mobile communication technology.
When supporting traditional voice service, the all-IP mobile communication network
has the problem of how to simultaneously support several kinds of vocoders with lower
cost, that is, the problem of supporting so-called TrFO (Transcoder Free Operation)
and RTO (Remote Transcoder Opearation).
[0008] TrFO means making the network negotiate the encoding and decoding type and the mode
of the vocoder before call establishment through some out-band negotiation mechanism.
Through negotiation, calls among the mobile subscribers do not need to pass through
the vocoder at network side, thus improving the voice quality, saving expensive vocoder
resource and the followed power consumption.
[0009] RTO is a special case of TrFO, and since the encoding and decoding methods of both
sides of communication are not consistent during out-band negotiation, it is necessary
that there is a vocoder at network side to transform the code type of one side of
communication into the code type of the other side. The main difference between RTO
and TDM circuit transmission networks is that: in TDM network, there are two encoding
and decoding transformations at network side, while only one in RTO network. Taking
RTO as an example here, MS1 uses EVRC while MS2 uses QCELP-13k, during the call between
MS1 and MS2, the voice of MS1 subscriber arrives at the ears of MS2 subscriber through
the following process: firstly, MS1 sends the encoded EVRC compressed voice frames
to network side 1 through radio uplink; network side 1 directly switches the received
EVRC voice frames and sends them to network side 2; network side 2 receives the switched
EVRC compressed voice frames and transforms them into QCELP-13k compressed voice frames
by a vocoder, and sends them to MS2 through radio downlink.
[0010] Taking CDMA2000 LMSD (Legacy Mobile Station Domain) as an example, out-band negotiation
of TrFO is implemented through signaling negotiation between access network and MSCe.
Since CDMA2000 LMSD applies IP switching technology, network side can directly make
the compressed voice data encoded by the UE be RTP (Real-Time Transport Protocol)
packets and transfer them through IP network, and there is no need to transform encoding
and decoding scheme to PCM and to be transmit through TDM circuit.
[0011] Taking EVRC as an example, as inter alia disclosed in 3
rd Generation Partnership Project 2, "3GPP2", "Enhanced Variable Rate Codec, Speech
Service Option 3 and 68 for Wideband Spread Spectrum Digital Systems", 3GPP2 C.S0014-B,
Version 1.0, May 2006, the maximal transmission rate of EVRC is 8kpbs (the transmission
rate of full-rate frame), and EVRC also has lots of half-rate frames and 1/8-rate
frames. Statistical analysis shows that, in EVRC call, the proportion of full-rate
frames is averagely around 30%, and its transmission rate is 22 bytes/20ms frame;
the proportion of half-rate frames is averagely around 30%, and its transmission rate
is 10 bytes/20ms frame; the proportion of 1/8-rate frames is averagely around 40%,
and its transmission rate is 2 bytes/20ms frame. In addition, since RTP transmission
supports multi-frame packing function, EVRC can be packed to be transmitted in the
network to save the overhead of IP header. Taking packing 3 EVRC frames into a RTP
message for example, the average transmission rate of EVRC in the nework is 11.7kbps
with the overhead of IP header. While the transmission rate of one-path voice PCM
code stream in network is 64kbps in the previous TDM circuit transmission, thus the
bandwidth used to transmit compressed voice in all-IP saves about (1-11.7/64)=81.7%
of PCM code stream bandwidth in TDM circuit mode. This example illustrates that, TrFO
can save a lot of network bandwidth.
[0012] However, TrFO has some problems in the practical applications. For example, suppose
MS1 and MS2 are in a TrFO call, if the quality of radio link is not good enough, there
is possibility that network side 1 cannot correctly receive and analyze the content
of some frames sent by MS 1 to network side 1 through uplink, that is, frame error
in the air. These frames that cannot be analyzed can be smoothly processed by the
vocoder at the network side in TDM circuit transmission network, while in TrFO technology,
since there is no participation of vocoder, network side 1 only can fill these frame
errors with to-be-compensated frames specified in protocol (for example, in EVRC,
half-rate or full-rate frames with all bits being "0" are defined as to-be-compensated
frames, and so on) and switch them to network side 2 which sends these to-be-compensated
frames specified by protocol to MS2. Meanwhile, because of the characteristic of IP
network transmission, there is possibility of frame loss or frame jittering during
the process of the voice frames at network side 1 arriving at network side 2 through
network, at this time, if network side 2 does not receive frames from network side
1 in the specified time, it will fill the frame errors with to-be-compensated frames
according to the protocol and send them to MS2. After these to-be-compensated frames
due to the quality of radio link and the quality of network transmission arrive at
MS2, there would be no problem if MS2 performs voice compensation for these frames;
however, a lot of experiments show that, most UEs will not perform voice compensation
for these to-be-compensated frames. Therefore, these to-be-compensated frames will
have a big influence on the whole voice quality of TrFO.
[0013] RTO technology also has the same problem. Since RTO call uses vocoder at network
side, suppose MS1 and MS2 are in RTO call, some of the frames sent by MS1 to network
side 1 through uplink may have errors if the quality of radio link is not good enough,
network side 1 can uses the vocoder at network side to perform voice compensation
for these frame errors. However, when the compensated voice frames arrives at network
side 2, there is still possibility of frame loss and frame jittering due to the problem
of network transmission quality, at this time, network side 2 will fill the frame
errors with to-be-compensated frames specified in protocol and send them to MS2. Therefore,
these to-be-compensated frames will have big influence on the whole voice quality
of RTO if voice compensation cannot be performed effectively for these frames by MS2.
[0014] To sum up, when the environment of radio link is good and the quality of network
transmission is ideal, TrFO and RTO reducing the times of encoding and decoding by
the vocoder at network side will improve the voice quality, but they cannot use the
vocoder at network side for voice compensation as the original circuit-switched mobile
communication system does when the environment of radio link is not good and the quality
of network transmission is poor. At this time, voice compensation totally relies on
the vocoder at the UE side. To date, the UEs produced by various manufacturers in
the market have different solutions regarding to whether it needs to compensate the
received to-be-compensated voice frames or not, and thus the voice qualities of TrFO
and RTO severely rely on the compensation performance of the vocoder in the UE and
on whether the vocoder compensates all kinds of to-be-compensated voice frames or
not, which will significantly affect the whole voice quality of TrFO and RTO.
[0015] Practice proves that, if one of consecutive full-rate frames is damaged or lost,
what the UE receives is to-be-compensated frames in the case of TrFO or RTO. The voice
quality when the UE processes the to-be-compensated frames is evidently worse than
the voice quality in the case that there is vocoder at network side in TDM circuit
transmission network, there are phenomena of words swallowing, trilling and discontinuous
voice in the former case. For the UEs having different vocoders, the degrees of how
good or poor the voice quality is are not the same.
Summary of the Invention
[0016] The present invention offers a method and system for realizing voice compensation
in a mobile communication network to solve the above disadvantages and to approximately
compensate the voice in the condition that the transmission quality is poor and the
device at network side does not use or partially uses vocoder, thus improving the
quality of whole voice.
[0017] The technique scheme applied in the present invention is:
[0018] A method for realizing voice compensation in mobile communication network, as set
forth in independent claim 1.
[0019] The present invention also offers a system for realizing voice compensation in mobile
communication network, as set forth in independent claim 3.
[0020] Preferred embodiments are set forth in the dependent claims.
[0021] The system and method of the present invention can effectively solve the problem
that the voice quality makes the ears uncomfortable, including discontinuous phenomenon,
trilling and swallowing words during the call, due to the bad radio link environment
or poor network transmission quality when there is no vocoder at the network side
participates or vocoder at the network side partially participates the call. The scheme
of the present invention implements voice compensation in the device at the network
side to effectively reduce the dependency of the call on the user terminal and the
vocoder's performance, thus satisfying all kinds of voice quality requirements by
the user terminals.
Brief Description of Drawings
[0022]
FIG. 1 is a flow chart of specifically implementing the method for voice compensation
in accordance with the present invention;
FIG. 2 is an illustration of the system for specifically implementing voice compensation
in accordance with the present invention;
FIG. 3 is a flow chart in accordance with the embodiment 1 of the present invention;
FIG 4 is a flow chart in accordance with the embodiment 2 of the present invention;
FIG 5 is a flow chart in accordance with the embodiment 3 of the present invention.
Preferred Embodiments of the Present Invention
[0023] The present invention will be described in further detail with reference to the accompanying
figures and embodiments.
[0024] The main idea of the present invention is: during the call, full-rate frames and
half-rate frames gives the greatest contribution to the voice, if there are full-rate
frames or half-rate frames damaged or lost, the voice quality will be easily affected.
A lot of experiments show that, the loss or damage of one or more full-rate frames
will cause discontinuous voice, swallowing words, especially in the condition of consecutive
full-half-rate frames; and the loss or damage of one or more half-rate frames will
cause trilling especially in the condition of consecutive full-half-rate frames, thus
making the ears uncomfortable, and the degree of uncomfortable feeling relies on the
encoding and decoding performance of the vocoder of the UE. Therefore, the objective
of the present invention is to mainly compensate full-rate frames or half-rate ones.
[0025] The present invention offers a method for realizing voice compensation in mobile
communication network to apply in the condition that the radio environment is bad
or transmission quality is poor, and the network side does not use or partially uses
vocoder (such as TrFO or RTO), as shown in FIG. 1, and the method comprises the following
steps:
Step 1: The device at the network side judges the forward frame received from the
network side or to be sent and processed is an invalid frame or not at each time of
processing the forward voice frames; or the network side judges the backward frame
received from the UE or to be sent and processed is an invalid frame or not at each
time of processing the backward voice frames;
If it is invalid frame, the process proceeds to step 2;
Otherwise, normally process and output the voice frames.
Said "invalid frames" means the following kinds of frames:
Blank frames, deleted frames, or other undefined-rate frames in protocol;
Or frames not received at the specified time of processing the frames (such as loss
frames or frames arrived later due to jittering);
Or frames needed to be compensated after being received by the vocoder specified in
the protocol.
Step 2: the device at the network side judges whether the voice compensation should
be performed for the invalid frame. The basis of judgment is whether the invalid frame
is the frame in the condition of non-1/8-rate :
If the invalid frame is a frame in the condition of non-1/8-rate, the invalid frame
will have a relatively big influence on the voice quality, the process proceeds to
step 3;
If the invalid frame is a frame in the condition of 1/8-rate, the invalid frame will
have little influence on the voice quality and does not need to be compensated, the
invalid frame is normally processed and output.
[0026] The method for judging whether the invalid frame is a frame in the condition of non-1/8-rate
is as follows:
The device at the network side judges whether the last valid frame is a 1/8-rate frame
or not, if the last valid frame is a non-1/8-rate frame, the invalid frame is a frame
in the condition of non-1/8-rate; otherwise, the invalid frame is a frame in the condition
of 1/8-rate;
If the device at the network side judges each forward voice frame in step 1, it judges
the last valid frame before the forward voice frame in this step; if the device at
the network side judges each backward voice frame in step 1, it judges the last valid
frame before the backward voice frame in this step.
[0027] Said "valid frame" means that the frame can be normally encoded and decoded by the
vocoder during the call, that is, those frames other than the invalid frames are called
valid frames.
[0028] Said "last valid frame" means the valid frame received or to be sent at the last
time of processing frames; if the frame received or to be sent at the last time of
processing frames is an invalid frame, it means the valid frame received or to be
sent at the time just before the last one of processing frames, and so on.
[0029] Step 3: the device at the network side judges whether the frame distance between
the invalid frame and its last valid frame is less than or equals to the threshold
of voice compensation:
If yes, the process proceeds to step 4;
Otherwise, there is no voice compensation, and the invalid frame is normally processed
and output.
[0030] Said compensation threshold depends on the compensation effect and performance of
the mobile communication system, the compensation threshold to achieve the optimal
compensation effect can be obtained according to the voice quality by comparing the
results of several experiments; For example, the compensation threshold is set as
6, 6 consecutive invalid frames are compensated; while if the compensation threshold
is set as 2, only two invalid frames are compensated, and the third one of the consecutive
frames will not be compensated.
[0031] For the "frame distance", for example, the frame distance between frame A and frame
B means the number of frames between frame A and frame B plus 1 for a group of sequentially
arrived frames. For example, for a group of sequentially arrived frames, frame a,
frame b, frame c, frame d ... ..., the frame distance between frame a and frame d
is 3.
[0032] Step 4: the device at the network side performs voice frame compensation for the
invalid frame, uses the compensated voice frame to replace the invalid frame to become
the voice frame to be processed and output at this time. The voice frame compensation
methods applied by the device at the network side comprise one of the following: valid
frame copying method, 1/4-rate frame filling method, simulation approximating method,
and so on.
[0033] Valid frame copying method: the last valid frame is used to replace the current invalid
frame;
1/4-rate frame filling method: this method can only be applied in EVRC encoded and
decoded voice call; wherein a 1/4-rate frame with any content is used to replace the
current invalid frame to be compensated;
Simulation approximating method: a frame is simulated using the rate and content of
the last valid frame and the frame distance between the current invalid frame and
its last valid frame according to the discipline obtained from simulation, and the
frame obtained from simulation is used to replace the current invalid frame.
[0034] After compensation, the compensated voice frame is normally processed and output.
[0035] The present invention offers a system for realizing voice compensation in mobile
communication network, the system is set in the device at the network side and is
applied in the condition that the radio environment is bad or transmission quality
is poor and the device at the network side does not use or partially use vocoder,
as shown in FIG. 2, and the system comprises:
an invalid frame detection unit for judging whether the forward or backward voice
frame received or to be sent by the device at the network side is an invalid frame
or not at each time of processing frames; sending the invalid frame to the voice compensation
unit and sending the valid frame to the unit for processing the voice frames in the
device at the network side;
a voice compensation unit, comprising:
a voice compensation judgment unit for receiving the invalid frame sent by the invalid
frame detection unit, sending the invalid frame in the condition of non-1/8-rate whose
frame distance from the last valid frame is less than or equals to compensation threshold
to the voice compensation process unit, and sending other invalid frame to the unit
for processing voice frames in the device at the network side;
a voice compensation process unit for receiving invalid frames sent by the voice compensation
judgment unit and performing voice compensation for them; that is, performing one
of the following processes:
using the last valid frame to replace the current invalid frame;
using a 1/4-rate frame with any content to replace the current invalid frame; or using
the frame obtained through simulation to replace the current invalid frame.
[0036] The compensated voice frames are sent to the unit for processing voice frames in
the device at the network side.
[0037] Said device at the network side can be one of the following: a base station, a base
station controller, a radio network controller or a mobile switching center.
[0038] The present invention can be used for voice call in which the device at the network
side does not use or partially uses vocoder for voice compensation, including the
mobile communication system applying TrFO technology, RTO technology or TFO (Tandem
Free Operation) technology. The present invention also can be used in a mobile communication
system of CDMA2000, WCDMA (Wideband-Code Division Multiple Access) and TDS-CDMA (Time
Division Synchronization- Code Division Multiple Access).
[0039] In the following, the present invention will be described in further detail with
three embodiments.
Embodiment 1: Using valid frame copying method to implement voice compensation.
[0040] The method for voice compensation applied in this embodiment is valid frame copying
method, and the threshold of frame distance of voice compensation in this embodiment
is 1, that is, voice compensation is performed only for the first invalid frame after
the valid frames in the condition of full-rate, while voice compensation is not performed
for the continuously arrived invalid frames after the first invalid frame; As shown
in FIG. 3, the specific steps which should be performed are as follows:
101: The device at the network side judges the received forward voice frames from
the network side at each time of processing the forward voice frame:
If the forward voice frame arrived at this time is an invalid frame, the process proceeds
to step 102;
If the forward voice frame arrived at this time is a valid frame, the process directly
proceeds to step 104;
102: The device at the network side judges the last arrived frame;
If the last frame is also an invalid frame, there is no special processing and the
process directly proceeds to step 104;
If the last frame is non-full-rate frame, there is no special processing and the process
directly proceeds to step 104;
If the last frame is a full-rate frame, the process proceeds to step 103; please note
that the full-rate frame here is a valid frame.
103: The current invalid frame is discarded, the last arrived frame, that is, said
full-rate frame, is used to replace this invalid frame; the process proceeds to step
104;
104: The current forward voice frame is normally processed and output.
[0041] Although this embodiment only describes the steps of judging and compensating the
forward voice frames from the network side by the device at the network side, it is
also applicable for the steps of judging and compensating the backward voice frames
from the UE by the device at the network side, which will not be described in detail
here.
Embodiment 2: Using 1/4-rate frame filling method to implement voice compensation.
[0042] This embodiment can be applied for EVRC encoded and decoded calls, and the legal
encoding format of EVRC does not include 1/4-rate frames. A lot of experiments show
that, vocoder of each UE will perform voice compensation after receiving EVRC encoded
and decoded 1/4-rate frames. As shown in FIG 4, the specific steps of this embodiment
are as follows:
201: The device at the network side judges the rate of the received forward voice
frames from the network side at each time of processing the forward voice frame:
If the forward voice frame arrived at this time is an invalid frame, the process proceeds
to step 202;
If the forward voice frame arrived at this time is a normal voice frame, the process
directly proceeds to step 205;
202: The device at the network side judges the rate of the last valid frame:
If the last valid frame is a full-rate frame, the process proceeds to step 203;
If the last valid frame is non-full-rate frame, there is no special processing and
the process directly proceeds to step 205;
203: The device determines the frame distance between the last valid frame and the
current invalid frame:
If the frame distance is less than or equals to the predetermined compensation threshold,
the process proceeds to step 204;
Otherwise, the process proceeds to step 205;
204: The current invalid frame is discarded and a 1/4-rate frame with any content
is used to replace this invalid frame. This 1/4-rate frame is used as the forward
voice frame at this time. The process proceeds to step 205;
205: the current forward voice frame is normally processed and output.
[0043] From the above steps, it can be seen that the main idea of this embodiment is replacing
a group of consecutive frames right after the full-rate frames with 1/4-rate frames;
For each invalid frame whose frame distance from its last full-rate valid frame is
less than or equals to the predetermined threshold, it is replaced with a 1/4-rate
frame; No extra voice compensation is performed for those whose frame distance is
greater than the threshold; that is, if the number of consecutive invalid frames right
after the full-rate valid frame is greater than the maximal threshold, no extra voice
compensation will be performed for those ones which exceed the threshold; said maximal
threshold of the number of invalid frames is the compensation threshold. In practical
applications, the compensation threshold of this method can be set as infinite, that
is, all consecutive invalid frames right after the full-rate frames will be replaced
with 1/4-rate frames.
[0044] Although this embodiment only describes the steps of judging and compensating the
forward voice frames from the network side by the device at the network side, it is
also applicable for the steps of judging and compensating the backward voice frames
from UE by the device at the network side, which will not be described here.
Embodiment 3: Using simulation approximating method to implement voice compensation.
[0045] In this embodiment, a large amount of full-rate voice data are statistically inducted
according to the previous actual situations to obtain the approximation rule of the
content of frames and rate change; when the invalid frames are compensated, a frame
can be simulated and used to replace the rate and content of the invalid frame according
to the approximation rule and by using the content and rate of the last validate frame
and the frame distance between the invalid frame and the last valid frame; in this
specification, the frames obtained through simulation are called as pseudo-full-rate
frames. The predetermined compensation threshold in this embodiment is 6. As shown
in FIG. 5, the specific steps of this embodiment are as follows:
301: The device at the network side judges the rate of the received forward voice
frames from the network side at each time of processing the forward voice frame:
If the frame arrived at this time is an invalid frame, the process proceeds to step
302;
If the frame arrived at this time is a normal voice frame, the process directly proceeds
to step 305;
302: The device judges the rate of the saved last valid frame:
If the last valid frame is full-rate, the process proceeds to step 303;
If the last valid frame is non-full-rate frame, there is no special processing and
the process directly proceeds to step 305;
303: The device determines the frame distance between the last valid frame and the
current invalid frame:
If the frame distance is less than or equals to 6, the process proceeds to step 304;
Otherwise, the process proceeds to step 305;
304: This invalid frame is discarded, a pseudo full-rate frame is simulated according
to the approximation rule obtained through statistical induction by using the content
of last valid frame and the frame distance between the last valid frame and the current
invalid frame as parameters, the simulated pseudo full-rate frame is used to replace
this invalid frame; the pseudo full-rate frame is used as the forward voice frame
at this time; the process proceeds to step 305;
305: The current forward voice frame is normally processed and output.
[0046] From the above steps, it can be seen that the main idea of this embodiment is replacing
the consecutive invalid frames right after the full-rate frames with simulated voice
frames, in simulation, the 6 consecutive invalid frames right after the full-rate
frames can be compensated through statistical discipline according to the content
of the full-rate frames and the frame distance between the invalid frame and the full-rate
frame.
[0047] Although this embodiment only describes the steps of judging and compensating the
forward voice frames from the network side by the device at the network side, it is
also applicable for the steps of judging and compensating the backward voice frames
from the UE by the device at the network side, which will not be described here.
[0048] The above three embodiments have their own advantages, but for voice quality, the
voice quality obtained through simulation approximating method is the best, and moreover,
the simulation approximating method can compensate several deleted frames in the condition
of consecutive full-rate frames, and its overhead is not heavy, only the content of
the latest full-rate frame should be saved.
[0049] The above three embodiments mainly compensate the frames in the condition of full-rate;
in practical applications, it may be supposed that voice compensation will be performed
when the last valid frame is a full-rate or half-rate frame. In addition, in practical
applications, the compensation threshold can be set according to the practical situations.
Industrial Applicability
[0050] The present invention offers a system and method for realizing voice compensation
at the network side to solve the problem that the whole voice quality is poor and
makes human ears uncomfortable because vocoder is not used or partially used in the
network side for voice quality compensation and linear prediction so as to make the
voice quality largely rely on whether the vocoder in the UE compensates some to-be-compensated
frames and on the compensation performance. When the radio environment is bad or the
network transmission quality is relatively poor, the technical scheme of the present
invention can relatively compensate the voice when vocoder is not used or partially
used at the network side to reduce the uncomfortable feelings of ears due to swallowing
words, trilling and discontinuous voice, to promote the whole voice quality, and to
reduce the dependency of calls on the performance of UE and its vocoder.