Apparatus and method realizing improved concepts for TCX LTP

• US 9,997,163 B2
• Filed: 12/18/2015
• Issued: 06/12/2018
• Est. Priority Date: 06/21/2013
• Status: Active Grant

• Current Assignee: Fraunhofer Gesellschaft Zur Foerderung Der Angewandten Forsching E.V.
• Original Assignee: Fraunhofer Gesellschaft Zur Foerderung Der Angewandten Forsching E.V.
• Inventors: Schnabel, Michael, Markovic, Goran, Sperschneider, Ralph, Lecomte, Jeremie, Helmrich, Christian
• Primary Examiner(s): Sirjani, Fariba
• Application Number:
  US14/973,727
• Publication Number:
  US 20160104489A1
• Time in Patent Office: 907 Days
• Field of Search: None
• CPC Class Codes: G10L 19/002 Dynamic bit allocation for perceptual audio coders G10L19/032
• CPC Class Codes: G10L 19/005 Correction of errors induced by the transmission channel, if related to the coding algorithm
• CPC Class Codes: G10L 19/012 Comfort noise or silence coding
• CPC Class Codes: G10L 19/0212 using orthogonal transformation
• CPC Class Codes: G10L 19/06 Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
• CPC Class Codes: G10L 19/07 Line spectrum pair [LSP] vocoders
• CPC Class Codes: G10L 19/083 the excitation function being an excitation gain G10L25/90 takes precedence
• CPC Class Codes: G10L 19/09 Long term prediction, i.e. removing periodical redundancies, e.g. by using adaptive codebook or pitch predictor
• CPC Class Codes: G10L 19/12 the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
• CPC Class Codes: G10L 19/22 Mode decision, i.e. based on audio signal content versus external parameters
• CPC Class Codes: G10L 2019/0002 Codebook adaptations
• CPC Class Codes: G10L 2019/0011 Long term prediction filters, i.e. pitch estimation
• CPC Class Codes: G10L 2019/0016 Codebook for LPC parameters

Abstract

An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal is provided. The apparatus includes a receiving interface, a delay buffer and a sample processor for processing the selected audio signal samples to obtain reconstructed audio signal samples of the reconstructed audio signal. The sample selector is configured to select, if a current frame is received by the receiving interface and if the current frame being received by the receiving interface is not corrupted, the plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer depending on a pitch lag information being included by the current frame.

Claims

1. An apparatus for decoding an encoded audio signal to acquire a reconstructed audio signal, wherein the apparatus is configured to receive a plurality of frames, and wherein the apparatus comprises:

an inverse modified discrete cosine transform module for decoding the plurality of frames by conducting an inverse modified discrete cosine transform to acquire audio signal samples of the decoded audio signal, and

a long-term prediction unit for conducting long-term prediction, comprising;

a delay buffer for storing the audio signal samples of the decoded audio signal,

a sample selector for selecting a plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer, and

a sample processor for processing the selected audio signal samples to acquire reconstructed audio signal samples of the reconstructed audio signal,

wherein the sample selector is configured to select, if a current frame is received by the apparatus and if the current frame being received by the apparatus is not corrupted, the plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer depending on a pitch lag information being comprised by the current frame, and

wherein the sample selector is configured to select, if the current frame is not received by the apparatus or if the current frame being received by the apparatus is corrupted, the plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer depending on a pitch lag information being comprised by another frame being received previously by the apparatus,

wherein the sample selector is configured to acquire the reconstructed audio signal samples by rescaling the selected audio signal samples depending on a modified gain, wherein the modified gain is defined according to the formula;

gain=gain_past*damping;

wherein gain is the modified gain,

wherein the sample selector is configured to set gain_past to gain after gain has been calculated, and

wherein damping is a real value with 0≤

damping≤

1.

2. An apparatus according to claim 1,

wherein the sample processor is configured to acquire the reconstructed audio signal samples, if the current frame is received by the apparatus and if the current frame being received by the apparatus is not corrupted, by rescaling the selected audio signal samples depending on a gain information being comprised by the current frame, and

wherein the sample selector is configured to acquire the reconstructed audio signal samples, if the current frame is not received by the apparatus or if the current frame being received by the apparatus is corrupted, by rescaling the selected audio signal samples depending on the gain information being comprised by said another frame being received previously by the apparatus.

3. An apparatus according to claim 2,

wherein the sample processor is configured to acquire the reconstructed audio signal samples, if the current frame is received by the apparatus and if the current frame being received by the apparatus is not corrupted, by multiplying the selected audio signal samples and a value depending on the gain information being comprised by the current frame, and

wherein the sample selector is configured to acquire the reconstructed audio signal samples, if the current frame is not received by the apparatus or if the current frame being received by the apparatus is corrupted, by multiplying the selected audio signal samples and a value depending on the gain information being comprised by said another frame being received previously by the apparatus.

4. An apparatus according to claim 1, wherein the sample processor is configured to store the reconstructed audio signal samples into the delay buffer.

5. An apparatus according to claim 4, wherein the sample processor is configured to store the reconstructed audio signal samples into the delay buffer before a further frame is received by the apparatus.

6. An apparatus according to claim 4, wherein the sample processor is configured to store the reconstructed audio signal samples into the delay buffer after a further frame is received by the apparatus.

7. An apparatus according to claim 1, wherein the sample processor is configured to rescale the selected audio signal samples depending on the gain information to acquire rescaled audio signal samples and by combining the rescaled audio signal samples with input audio signal samples to acquire the processed audio signal samples.

8. An apparatus according to claim 7,

wherein the sample processor is configured to store the processed audio signal samples, indicating the combination of the rescaled audio signal samples and the input audio signal samples, into the delay buffer, and to not store the rescaled audio signal samples into the delay buffer, if the current frame is received by the apparatus and if the current frame being received by the apparatus is not corrupted, and

wherein the sample processor is configured to store the rescaled audio signal samples into the delay buffer and to not store the processed audio signal samples into the delay buffer, if the current frame is not received by the apparatus or if the current frame being received by the apparatus is corrupted.

9. An apparatus according to claim 7, wherein the sample processor is configured to store the processed audio signal samples into the delay buffer, if the current frame is not received by the apparatus or if the current frame being received by the apparatus is corrupted.

10. An apparatus according to claim 1, wherein the sample selector is configured to calculate the modified gain.

11. An apparatus according to claim 1, wherein the modified gain is set to zero, if at least a predefined number of frames have not been received by the apparatus since a frame last has been received by the apparatus.

12. A method for decoding an encoded audio signal to acquire a reconstructed audio signal, wherein the method comprises:

receiving a plurality of frames,

decoding the plurality of frames by conducting an inverse modified discrete cosine transform to acquire audio signal samples of the decoded audio signal,

conducting long-term prediction by

storing the audio signal samples of the decoded audio signal,

selecting a plurality of selected audio signal samples from the audio signal samples being stored in a delay buffer, and

processing the selected audio signal samples to acquire reconstructed audio signal samples of the reconstructed audio signal,

wherein, if a current frame is received and if the current frame being received is not corrupted, selecting the plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer is conducted depending on a pitch lag information being comprised by the current frame, and

wherein, if the current frame is not received or if the current frame being received is corrupted, selecting the plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer is conducted depending on a pitch lag information being comprised by another frame being received previously,

wherein the method further comprises rescaling the selected audio signal samples depending on a modified gain, wherein the modified gain is defined according to the formula;

gain=gain_past*damping;

wherein gain is the modified gain,

wherein gain_past is set to gain after gain has been calculated, and

wherein damping is a real value with 0≤

damping≤

1.

13. A non-transitory computer-readable medium comprising a computer program for implementing the method of claim 12 when being executed on a computer or signal processor.