System and method for reducing tandeming effects in a communication system
View Patent ↗The present disclosure is directed towards a system and method for reducing tandeming effects in a communications system. The method may include receiving, at a speech decoder, an input bitstream associated with an incoming initial speech signal from a speech encoder. The method may further include determining whether or not coding is required and if coding is required, modifying an excitation signal associated with the bitstream. The method may also include providing the modified excitation signal to an adaptive encoder.
1. A computer-implemented method for reducing tandeming effects in a communications system comprising:
receiving an incoming initial audible audio speech signal from a user at a speech encoder;
receiving, at a speech decoder, an input bitstream associated with the incoming initial audible audio speech signal from the speech encoder;
determining, at the speech decoder, a decoded signal based on the received input bitstream;
determining whether or not coding is required, wherein determining whether or not coding is required comprises comparing an excitation signal constructed using the decoded signal to a previous optimized final version of the excitation signal from a previous frame;
upon determining that coding is required,
modifying the excitation signal associated with the input bitstream;
providing the modified excitation signal to an adaptive encoder;
calculating a new excitation signal e 1 ( n ) for a new speech signal s(n) after Voice Quality Assurance (“VQA”) processing has been performed, wherein VQA processing is performed on the decoded signal prior to providing the signal to the adaptive encoder;
calculating a total excitation signal e 2 ( n ) for the decoded signal sp(n) before VQA; and
deciding whether the decoded signal from the speech decoder is copied to an output of the adaptive encoder using an adaptive coding algorithm, by calculating a distance over signal (“DOS”) ratio “R” between the new excitation signal e 1 ( n ) and the total excitation signal e 2 ( n ); and
encoding the modified excitation signal at the adaptive encoder, and transmitting the encoded modified excitation signal with the decoded signal based on deciding that the decoded signal is to be copied.
2. The method of claim 1 , further comprising:
decoding T-milliseconds codec frames of the input bitstream that was encoded by a code excited linear prediction (“CELP”) based encoder at a rate of S kilobits/second for an adaptive encoder.
3. The method of claim 1 , further comprising:
calculating a total excitation signal u(n) by adding an adaptive and a fixed codebook vector, each scaled by a respective gain.
4. The method of claim 3 , further comprising:
saving the total excitation signal u(n) at an adaptive encoder memory, wherein at least one of the adaptive codebook vector and the fixed codebook vector include speech frames and discontinuous transmission (“DTX”) frames based on a CELP-based standard.
5. The method of claim 2 , wherein the adaptive encoder includes an adaptive encoder memory configured to store a defined data structure.
6. The method of claim 1 , further comprising:
performing adaptive excitation synchronization based upon, at least in part, a final decision flag decision generated by the adaptive coding algorithm.
7. The method of claim 2 , further comprising:
disabling a post-processing option including a high-pass filter in a partial decoder, wherein disabling is configured to avoid codec tandeming.
8. A system for reducing tandeming effects in a communications system, the system including at least one processor configured to perform operations comprising:
receiving an incoming initial audible audio speech signal from a user at a speech encoder;
receiving, at a speech decoder, an input bitstream associated with the incoming initial audible audio speech signal from the speech encoder;
determining, at the speech decoder, a decoded signal based on the received input bitstream;
determining whether or not coding is required, wherein determining whether or not coding is required comprises comparing an excitation signal constructed using the decoded signal to a previous optimized final version of the excitation signal from a previous frame;
upon determining that coding is required,
modifying the excitation signal associated with the input bitstream;
providing the modified excitation signal to an adaptive encoder;
calculating a new excitation signal e 1 ( n ) for a new speech signal s(n) after Voice Quality Assurance (“VQA”) processing has been performed, wherein VQA processing is performed on the decoded signal prior to providing the signal to the adaptive encoder;
calculating a total excitation signal e 2 ( n ) for the decoded signal sp(n) before VQA; and
deciding whether the decoded signal from the speech decoder is copied to an output of the adaptive encoder using an adaptive coding algorithm, by calculating a distance over signal (“DOS”) ratio “R” between the new excitation signal e 1 ( n ) and the total excitation signal e 2 ( n ); and
encoding the modified excitation signal at the adaptive encoder, and transmitting the encoded modified excitation signal with the decoded signal based on deciding that the decoded signal is to be copied.
9. The system of claim 8 , further comprising:
decoding T-milliseconds codec frames of the input bitstream that was encoded by a code excited linear prediction (“CELP”) based encoder at a rate of S kilobits/second for an adaptive encoder.
10. The system of claim 8 , further comprising:
calculating a total excitation signal u(n) by adding an adaptive and a fixed codebook vector, each scaled by a respective gain.
11. The system of claim 10 , further comprising:
saving the total excitation signal u(n) at an adaptive encoder memory, wherein at least one of the adaptive codebook vector and the fixed codebook vector include speech frames and discontinuous transmission (“DTX”) frames based on a CELP-based standard.