摘要:
A method and apparatus for frame classification and rate determination in voice transcoders. The apparatus includes a classifier input parameter preparation module that unpacks the bitstream from the source codec and selects the codec parameters to be used for classification, parameter buffers that store previous input and output parameters of previous frames, and a frame classification and rate decision module that uses the source codec parameters from the current frame and zero or more frames to determine the frame class, rate, and classification feature parameters for the destination codec. The classifier input parameter preparation module separates the bitstream code and unquantizes the sub-codes into the codec parameters. These codec parameters may include line spectral frequencies, pitch lag, pitch gains, fixed codebook gains, fixed codebook vectors, rate and frame energy. The frame classification and rate decision module comprises M sub-classifiers and a final decision module. The characteristics of the sub-classifiers are obtained by a classifier construction module, which comprises a training set generation module, a learning module and an evaluation module. The method includes preparing the classifier input parameters, constructing the frame and rate classifier and determining the frame class, rate decision and classification feature parameters for the destination codec using the intermediate parameters and bit rate of the source codec. Constructing the frame and rate classifier includes generating the training and test data and training and/or building the classifier.
摘要:
A variable rate compressed voice signal domain transcoder that transcodes a bitstream representing frames of data encoded according to a first voice compression standard to a bitstream representing frames of data according to a second voice compression standard; the second voice compression standard defines a variable-rate voice codec. The method includes unquantizing a bitstream into a first set of parameters compatible with a first compression standard. The first set of parameters in addition to external control commands are then used to determine the frame class and rate for the second compression standard. Next, the first set of parameters are transformed into a second set of parameters compatible with a second compression standard according to the frame-classification and rate determination decision without converting the first set of parameters to an analog or digital voice waveform representation. The transformation approaches can be varied and further optimized based on the characteristics of the pair of first compression standard and the second compression standard. Lastly, the second set of parameters is packed into a bitstream compatible with the second compression standard.
摘要:
A method for providing a multimedia service to a multimedia terminal includes establishing an audio link between the multimedia terminal and a server over an audio channel, and detecting one or more media capabilities of the multimedia terminal. The method also includes providing an application logic for the multimedia service, establishing a visual link between the multimedia terminal and the server over a video channel, providing an audio stream for the multimedia service over the audio link, and providing a visual stream for the multimedia service over the video link. The method further includes combining the video link and the audio link, and adjusting a transmission time of one or more packets in the visual stream to synchronize the visual stream with the audio stream.
摘要:
The present invention relates to converting media bitstreams across different networks in a media gateway without any algorithmic delay, and reduces the computation load within the transmission in the situation where the bandwidth of the outgoing network varies dynamically. A first embodiment of the present invention provides an apparatus and a method for a Reduced-Algorithmic-Delay Media Stream Unit Conversion module which is a light weight Simple Pass-Through operation. A second embodiment of the present invention provides an apparatus and a method for a Reduced-Algorithmic-Delay Video Rate Conversion. An alternative embodiment provides an apparatus and a method for a Smart Pass-Through Operation which involves switching between the Simple Pass-Through and the Rate Converter. The methods and apparatuses provided by the first and second embodiment can be used as a stand alone system, or as part of the module of the alternative embodiment.
摘要:
A method of removing a motion vector from a group of motion vectors used in an encoding process includes providing a list of motion vectors, selecting an initial motion vector from the list of motion vectors, and providing an intermediate motion vector using a motion vector refinement process. The motion vector refinement process uses, in part, the initial motion vector. The method also includes forming a region defined by one or more parameters associated with the initial motion vector and one or more parameters associated with the intermediate motion vector, selecting an additional motion vector from the list of motion vectors, determining that the additional motion vector points into the region, and modifying a state of the additional motion vector.
摘要:
An apparatus for transcoding an audio signal between a CELP-based coder and a hybrid coder includes a source bitstream unwrapper configured to receive a source bitstream, extract one or more CELP compression parameters from the source bitstream, and construct an audio signal vector from the source bitstream while maintaining the one or more extracted CELP compression parameters. The apparatus also includes a frame interpolator coupled to the source bitstream unwrapper and a compression parameter converter coupled to frame interpolator. The compression parameter converter is configured to calculate output compression parameters from at least one of the interpolated compression parameters or the one or more extracted CELP compression parameters. Additionally, the apparatus includes a destination bitstream wrapper coupled to the compression parameter converter and a mapping parameter tuner coupled to the frame interpolator. The mapping parameter tuner is configured to select one or more parameters for use by the compression parameter converter.
摘要:
A method and apparatus for DTMF detection and voice mixing in the code-excited linear prediction (CELP) parameter space, without fully decoding and reconstructing the speech signal. The apparatus includes a Dual Tone Multiplexed Frequency (DTMF) signal detection module and a multi-input mixing module. The DTMF signal detection module detects DTMF signals by computing characteristic features from the input CELP parameters and comparing them with known features of DTMF signals. The multi-input mixing module mixes multiple sets of input CELP parameters, that represent multiple voice signals, into a single set of CELP parameters. The mixing computation is performed by analyzing each set of input CELP parameters, determining the order of importance of the input sets, selecting a strategy for mixing the CELP parameters, and outputting the mixed CELP parameters. The method includes inputting one or more sets of CELP parameters and external commands, detecting DTMF tones, mixing multiple sets of CELP parameters and outputting the DTMF signal, if detected, and the mixed CELP parameters.
摘要:
Method and apparatus for trans-rating a bitstream of data through multi-rate voice coders converting a bitstream representing frames of data encoded according to a first voice compression method of a first rate to a second voice compression method according to a second rate. A trans-rating pair includes voice compression parameters mapping modules. The method of trans-rating includes either bit-unpacking or unquantization on an encoded packet at input site to obtain rate information and voice compression parameters according to the first rate voice compression method. The information of the first rate and the required output rate, namely a second rate type, in addition to external control commands, are then used to determine the converting strategy of the trans-rating pair. Next, at least some of the compression parameters of the first rate are passed through, or mapped, into compression parameters of the second rate compatible with the second rate voice compression method.
摘要:
A method for transcoding a CELP based compressed voice bitstream from source codec to destination codec. The method includes processing a source codec input CELP bitstream to unpack at least one or more CELP parameters from the input CELP bitstream and interpolating one or more of the plurality of unpacked CELP parameters from a source codec format to a destination codec format if a difference of one or more of a plurality of destination codec parameters including a frame size, a subframe size, and/or sampling rate of the destination codec format and one or more of a plurality of source codec parameters including a frame size, a subframe size, or sampling rate of the source codec format exist. The method includes encoding the one or more CELP parameters for the destination codec and processing a destination CELP bitstream by at least packing the one or more CELP parameters for the destination codec.
摘要:
A conferencing system is provided that utilizes both time domain signal mixing and direct signal fast transcoding. An exemplary embodiment of the present invention utilizes both time domain signal mixing and direct signal fast transcoding to process a bit-stream from a same channel during a conference.