Classifications

• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
  • G10L21/04—Time compression or expansion
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech 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
  • G10L19/04—Speech 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
  • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
  • G10L19/12—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech 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
  • G10L19/02—Speech 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 spectral analysis, e.g. transform vocoders or subband vocoders
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech 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
  • G10L19/02—Speech 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 spectral analysis, e.g. transform vocoders or subband vocoders
  • G10L19/0204—Speech 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 spectral analysis, e.g. transform vocoders or subband vocoders using subband decomposition
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech 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
  • G10L2019/0001—Codebooks
  • G10L2019/0012—Smoothing of parameters of the decoder interpolation

Definitions

• the present invention relates to audio processing and, in particular to an apparatus and method for processing an audio signal and for providing a higher temporal granularity for a Combined Unified Speech and Audio Codec (USAC).
• USAC as other audio codecs, exhibits a fixed frame size (USAC: 2048 samples/frame). Although there is the possibility to switch to a limited set of shorter transform sizes within one frame, the frame size still limits the temporal resolution of the complete system.
• the sampling rate is increased, leader to a shorter duration of one frame in time (e.g. milliseconds).
• this is not easily possible for the USAC codec:
• AAC Advanced Audio Coding
• SBR Spectral Band Replication
• MPEG Moving Picture Experts Group
• Both, ACELP and transform coder run usually at the same time within the same environment (i.e. frame size, sampling rate), and can be easily switched: usually, for clean speech signals, the ACELP tool is used, and for music, mixed signals the transform coder is used.
• the ACELP tool is at the same time limited to work only at comparably low sampling rates. For 24 kbit/s, a sampling rate of only 17075Hz is used. For higher sampling rates, the ACELP tool starts to drop significantly in performance.
• the transform coder as well as SBR and MPEG Surround however would benefit from a much higher sampling rate, for example 22050 Hz for the transform coder and 44100 Hz for SBR and MPEG Surround. So far, however, the ACELP tool limited the sampling rate of the complete system, leading to a suboptimal system in particular for music signals.
• the object of the present invention is to provide improved concepts for an apparatus and method for processing an audio signal.
• the object of the present invention is solved by an apparatus according to claim 1, a method according to claim 15, an apparatus according to claim 16, a method according to claim 18 and a computer program according to claim 19.
• the current USAC RM provides high coding performance over a large number of operating points, tanging from very low bitrates such as 8 kbit/s up to transparent quality at bitrates of 128 kbit/s and above.
• a combination of tools such as MPEG Surround, SBR, ACELP and traditional transform coders are used.
• Such a combination of tools of course requires a joint optimization process of the tool interoperation and a common environment, where these tools are placed.
• An apparatus for processing an audio signal comprises a signal processor and a configurator.
• the signal processor is adapted to receive a first audio signal frame having a first configurable number of samples of the audio signal.
• the signal processor is adapted to upsample the audio signal by a configurable upsampling factor to obtain a processed audio signal.
• the signal processor is adapted to output a second audio signal frame having a second configurable number of samples of the processed audio signal.
• the configurator is adapted to configure the signal processor based on configuration information such that the configurable upsampling factor is equal to a first upsampling value when a first ratio of the second configurable number of samples to the first configurable number of samples has a first ratio value.
• the configurator is adapted to configure the signal processor such that the configurable upsampling factor is equal to a different second upsampling value, when a different second ratio of the second configurable number of samples to the first configurable number of samples has a different second ratio value.
• the first or the second ratio value is not an integer value.
• a signal processor upsamples an audio signal to obtain a processed upsamplcd audio signal.
• the upsampling factor is configurable and can be a non-integer value.
• the configurability and the fact that the upsampling factor can be a non-integer value increases the flexibility of the apparatus.
• the configurable upsampling factor has a different second upsampling value.
• the apparatus is adapted to take a relationship between the upsampling factor and the ratio of the frame length (i.e. the number of samples) of the second and the first audio signal frame into account.
• the configurator is adapted to configure the signal processor such that the different second upsampling value is greater than the first upsampling value, when the second ratio of the second configurable number of samples to the first configurable number of samples is greater than the first ratio of the second configurable number of samples to the first configurable number of samples.
• a new operating mode (in the following called "extra setting") for the USAC codec is proposed, which enhances the performance of the system for mid-data rates, such as 24 kbit/s and 32 kbit/s. It was found that for these operating points, the temporal resolution of the current USAC reference codec is too low. It is therefore proposed to a) increase this temporal resolution by shortening the core-coder frame sizes without increasing the sampling rate for the core-coder, and further b) to increase the sampling rate for SBR and MPEG Surround without changing the frame size for these tools.
• the proposed extra setting greatly improves the flexibility of the system, since it allows the system including the AC LLP tool to be operated at higher sampling rates, such as 44.1 and 48 kHz. Since these sampling rates are typically requested in the marketplace, it is expected that this would help for the acceptance of the US AC codec.
• the new operating mode for the current MPEG Unified Speech and Audio Coding (USAC) work item increases the temporal flexibility of the whole codec, by increasing the temporal granularity of the complete audio codec. If (assuming that the second number of samples remained the same) the second ratio is greater than the first ratio, then the first configurable number of samples has been reduced, i.e. the frame size of the first audio signal frame has been shortened. This results in a higher temporal granularity, and all tools which operate in the frequency domain and which process the first audio signal frame can perform better. In such a high efficient operating mode, however, it is also desirable to increase the performance of tools which process the second audio signal frame comprising the upsampled audio signal.
• Such an increase in performance of these tools can be realized by a higher sampling rate of the upsampled audio signal, i.e. by increasing the upsampling factor for such an operating mode.
• tools exist, such as the ACE LP decoder in USAC, which do not operate in the frequency domain, which process the first audio signal frame and which operate best when the sampling rate of the (original) audio signal is relatively low.
• These tools benefit from a high upsampling factor, as this means that the sampling rate of the (original) audio signal is relatively low compared to the sampling rate of the upsampled audio signal.
• the new operating mode increases the temporal flexibility of the whole codec, by increasing the temporal granularity of the complete audio codec.
• the configurator is adapted to configure the signal processor such that the configurable upsampling factor is equal to the first ratio value when the first ratio of the second configurable number of samples to the first configurable number of samples has the first ratio value, and wherein the configurator is adapted to configure the signal processor such that the configurable upsampling factor is equal to the different second ratio value when the second ratio of the second configurable number of samples to the first configurable number of samples has the different second ratio value.
• the configurator is adapted to configure the signal processor such that the configurable upsampling factor is equal to 2 when the first ratio has the first ratio value, and wherein the configurator is adapted to configure the signal processor such that the configurable upsampling factor is equal to 8/3 when the second ratio has the different second ratio value.
• the configurator is adapted to configure the signal processor such that the first configurable number of samples is equal to 1024 and the second configurable number of samples is equal to 2048 when the first ratio has the first ratio value, and wherein the configurator is adapted to configure the signal processor such that that the first configurable number of samples is equal to 768 and the second configurable number of samples is equal to 2048 when the second ratio has the different second ratio value.
• the temporal granularity of the core-coder is increased by shrinking the core-coder frame size from 1024 to 768 samples.
• the temporal granularity of the core coder is increased by 4/3 while leaving the sampling rate constant: This allows the ACELP to run at an appropriate sampling frequency (Fs).
• a resampling of ratio 8/3 (so far: ratio 2) is applied, converting a core-coder frame of size 768 at 3/8 Fs to a output frame of size 2048 at Fs.
• This allows the SBR tool and an MPEG Surround Tool to be run at a traditionally high sampling rate (e.g. 44100 Hz).
• a traditionally high sampling rate e.g. 44100 Hz.
• the signal processor comprises a core decoder module for decoding the audio signal to obtain a preprocessed audio signal, an analysis filter bank having a number of analysis filter bank channels for transforming the first preprocessed audio signal from a time domain into a frequency domain to obtain a frequency-domain preprocessed audio signal comprising a plurality of subband signals, a subband generator for creating and adding additional subband signals for the frequency-domain preprocessed audio signal, and a synthesis filter bank having a number of synthesis filter bank channels for transforming the first preprocessed audio signal from the frequency domain into the time domain to obtain the processed audio signal.
• the configurator may be adapted to configure the signal processor by configuring the number of synthesis filter bank channels or the number of analysis filter bank channels such that the configurable upsainpling factor is equal to a third ratio of the number of synthesis filter bank channels to the number of analysis filter bank channels.
• the subband generator may be a Spectral Band Replicator being adapted to replicate subband signals of the preprocessed audio signal generator for creating the additional subband signals for the frequency-domain preprocessed audio signal.
• the signal processor may furthermore comprise an MPEG Surround decoder for decoding the preprocessed audio signal to obtain a preprocessed audio signal comprising stereo or surround channels.
• the subband generator may be adapted to feed the frequency-domain preprocessed audio signal into the MPEG Surround decoder after the additional subband signals for the frequency-domain preprocessed audio signal have been created and added to the frequency-domain preprocessed audio signal.
• the core decoder module may comprise a first core decoder and a second core decoder, wherein the first core decoder may be adapted to operate in a time domain and wherein the second core decoder may be adapted to operate in a frequency domain.
• the first core decoder may be an ACELP decoder and the second core decoder may be a FD transfom decoder or a TCX transform decoder.
• the super- frame size for the ACELP codec is reduced from 1024 to 768 samples. This could be done by combining 4 ACELP frames of size 192 (3 sub-frames of size 64) to one core-coder frame of size 768 (previously: 4 ACELP frames of size 256 were combined to a core-coder frame of size 1024). Another solution for reaching a core- coder frame size of 768 samples would be for example to combine 3 ACELP frames of size 256 (4 sub-frames of size 64).
• the configurator is adapted to configure the signal processor based on the configuration information indicating at least one of the first configurable number of samples of the audio signal or the second configurable number of samples of the processed audio signal.
• the configurator is adapted to configure the signal processor based on the configuration information, wherein the configuration information indicates the first configurable number o f samples of the audio signal and the second configurable number of samples of the processed audio signal, wherein the configuration information is a configuration index.
• an apparatus for processing an audio signal comprises a signal processor and a configurator.
• the signal processor is adapted to receive a first audio signal frame having a first configurable number of samples of the audio signal.
• the signal processor is adapted to downsample the audio signal by a configurable downsampling factor to obtain a processed audio signal. Furthermore, the signal processor is adapted to output a second audio signal frame having a second configurable number of samples of the processed audio signal.
• the configurator may be adapted to configure the signal processor based on configuration information such that the configurable downsampling factor is equal to a first downsampling value when a first ratio of the second configurable number of samples to the first configurable number of samples has a first ratio value.
• the configurator is adapted to configure the signal processor such that the configurable downsampling factor is equal to a different second downsampling value, when a different second ratio of the second configurable number of samples to the first configurable number of samples has a different second ratio value.
• the first or the second ratio value is not an integer value.
• Fig. 1 illustrates an apparatus for processing an audio signal according to an embodiment
• Fig. 2 illustrates an apparatus for processing an audio signal according to another embodiment
• Fig. 3 illustrates an upsampling process conducted by an apparatus according to an embodiment
• Fig. 4 illustrates an apparatus for processing an audio signal according to a further embodiment
• Fig. 5 a illustrates a core decoder module according to an embodiment
• Fig. 5b illustrates an apparatus for processing an audio signal according to the embodiment of Fig. 4 with a core decoder module according to Fig. 5a,
• Fig. 6a illustrates an ACELP super frame comprising 4 ACHLP frames
• Fig. 6b illustrates an ACELP super frame comprising 3 ACELP frames
• Fig. 7 a illustrates the default setting of US AC
• Fig. 7b illustrates an extra setting for US AC according to an embodiment
• Fig. 8a, 8b illustrate the results of a listening test according to MUSHRA methodology.
• Fig. 9 illustrates an apparatus for processing an audio signal according to an alternative embodiment.
• Fig. 1 illustrates an apparatus for processing an audio signal according to an embodiment.
• the apparatus comprises a signal processor 1 10 and a configurator 120.
• the signal processor 1 10 is adapted to receive a first audio signal frame 140 having a first configurable number of samples 145 of the audio signal Moreover, the signal processor 1 10 is adapted to upsample the audio signal by a configurable upsampling factor to obtain a processed audio signal. Furthermore, the signal processor is adapted to output a second audio signal frame 150 having a second configurable number of samples 155 of the processed audio signal.
• the configurator 120 is adapted to configure the signal processor 1 10 based on configuration information ci such that the configurable upsampling factor is equal to a first upsampling value when a first ratio of the second configurable number of samples to the first configurable number of samples has a first ratio value. Moreover, the configurator 120 is adapted to configure the signal processor 1 10 such that the configurable upsampling factor is equal to a different second upsampling value, when a different second ratio of the second configurable number of samples to the first configurable number of samples has a different second ratio value.
• the first or the second ratio value is not an integer value.
• An apparatus may for example be employed in the process of decoding.
• the configurator 120 may be adapted to configure the signal processor 1 10 such that the different second upsampling value is greater than the first different upsampling value, when the second ratio of the second configurable number of samples to the first configurable number of samples is greater than the first ratio of the second configurable number of samples to the first configurable number of samples.
• the configurator 120 is adapted to configure the signal processor 1 10 such that the configurable upsampling factor is equal to the first ratio value when the first ratio of the second configurable number of samples to the first configurable number of samples has the first ratio value, and wherein the configurator 120 is adapted to configure the signal processor 1 10 such that the configurable upsampling factor is equal to the different second ratio value when the second ratio of the second configurable number of samples to the first configurable number of samples has the different second ratio value.
• the configurator 120 is adapted to configure the signal processor 110 such that the configurable upsampling factor is equal to 2 when the first ratio has the first ratio value, and wherein the configurator 120 is adapted to configure the signal processor 1 10 such that the configurable upsampling factor is equal to 8/3 when the second ratio has the different second ratio value.
• the configurator 120 is adapted to configure the signal processor 1 10 such that the first configurable number of samples is equal to 1024 and the second configurable number of samples is equal to 2048 when the first ratio has the first ratio value, and wherein the configurator 120 is adapted to configure the signal processor 1 10 such that that the first configurable number of samples is equal to 768 and the second configurable number of samples is equal to 2048 when the second ratio has the different second ratio value.
• the configurator 120 is adapted to configure the signal processor 110 based on the configuration information ci, wherein the configuration information ci indicates the upsampling factor, the first configurable number of samples of the audio signal and the second configurable number of samples of the processed audio signal, wherein the configuration information is a configuration index.
• the following table illustrates an example for a configuration index as configuration information:
• Index indicates the configuration index
• coreCoderFrameLength indicates the first configurable number of samples of the audio signal
• sbrRatio indicates the upsampling factor
• outputFrameLength indicates the second configurable number of samples of the processed audio signal
• Fig. 2 illustrates an apparatus according to another embodiment.
• the apparatus comprises a signal processor 205 and a configurator 208.
• the signal processor 205 comprises a core decoder module 210, an analysis filter bank 220, a subband generator 230 and a synthesis filter bank 240.
• the core decoder module 210 is adapted to receive an audio signal asl . After receiving the audio signal asl , the core decoder module 210 decodes the audio signal to obtain a preprocessed audio signal as2. Then, the core decoder module 210 feeds the preprocessed 5 audio signal as2, being represented in a time domain, into the analysis filter bank 220.
• the analysis filter bank 220 is adapted to transform the preprocessed audio signal as2 from a time domain into a frequency domain to obtain a frequency-domain preprocessed audio signal as3 comprising a plurality of subband signals.
• the analysis filter bank 220 has a
• the number of analysis filter bank channels determines the number of subband signals that are generated from the time-domain preprocessed audio signal as2.
• the number of analysis filter bank channels may be set by setting the value of a configurable parameter cl .
• the analysis filter bank 220 may be configured to have 32 or 24
• the number of analysis filter bank channels may be set according to configuration information ci of a configurator 208.
• the analysis filter bank 220 After transforming the preprocessed audio signal as2 into the frequency domain, the analysis filter bank 220 feeds the frequency-domain preprocessed audio signal as3 into the subband generator 230.
• the subband generator 230 is adapted to create additional subband signals for the frequency-domain audio signal as3. Moreover, the subband generator 230 is adapted to modify the preprocessed frequency-domain audio signal as3 to obtain a modified frequency-domain audio signal as4 which comprises the subband signals of the
• the subband generator 230 preprocessed frequency-domain audio signal as3 and the created additional subband signals created by the subband generator 230.
• the number of additional subband signals that are generated by the subband generator 230 is configurable.
• the subband generator is a Spectral Band Replicator (SBR).
• SBR Spectral Band Replicator
• the synthesis filter bank 240 is adapted to transfonn the modified frequency-domain preprocessed audio signal as4 from a frequency domain into a time domain to obtain a time-domain processed audio signal as 5.
• the synthesis filter bank 240 has a configurable 35 number of synthesis filter bank channels (synthesis filter bank bands).
• the number of synthesis filter bank channels is configurable, in an embodiment, the number of synthesis filter bank channels may be set by setting the value of a configurable parameter c2.
• the synthesis filter bank 240 may be configured to have 64 synthesis filter bank channels.
• the configuration information ci of the configurator 208 may set the number of analysis filter bank channels.
• the number of subband channels of the modified frequency-domain preprocessed audio signal as4 is equal to the number of synthesis filter bank channels.
• the configurator 208 is adapted to configure the number of additional subband channels that are created by the subband generator 230.
• the configurator 208 may be adapted to configure the number of additional subband channels that are created by the subband generator 230 such that the number of synthesis filter bank channels c2, configured by the configurator 208, is equal to the number of subband channels of the preprocessed frequency-domain audio signal as3 plus the number of additional subband signals created by the subband generator 230.
• the number of synthesis filter bank channels is equal to the number of subband signals of the modified preprocessed frequency-domain audio signal as4.
• the upsampling factor u can be set to a number that is not an integer value.
• a Spectral Band Replicator is capable to generate an arbitrary number of additional subbands from the original subbands, wherein the ratio of the number of generated additional subbands to the number of already available subbands does not have to be an integer.
• a Spectral Band Replicator may conduct the following steps: In a first step, the Spectral Band Replicator replicates the number of subband signals by generating a number of additional subbands, wherein the number of generated additional subbands may be an integer multiple of the number of the already available subbands. For example, 24 (or, for example, 48) additional subband signals may be generated from 24 original subband signals of an audio signal (e.g. the total number of subband signals may be doubled or tripled).
• el l is equal to el 2
• the number c l 1 of available subband signals is equal to the number cl 2 of subband signals needed. No subband adjustment is required.
• the number cl 1 of available subband signals is greater than the number cl 2 of subband signals needed.
• the highest frequency subband signals might be deleted. For example, if 64 subband signals are available and if only 61 subband signals are needed, the three subband signals with the highest frequency might be discarded. If cl 2 is greater than el l , then the number cl 1 of available subband signals is smaller than the number el 2 of subband signals needed.
• additional subband signals might be generated by adding zero signals as additional subband signals, i.e. signals where the amplitude values of each subband sample are equal to zero.
• additional subband signals might be generated by adding pseudorandom subband signals as additional subband signals, i.e. subband signals where the values of each subband sample comprise pseudorandom data.
• additional subband signals might be generated by copying the sample values of the highest subband signal, or the highest suband signals, and to use them as sample values of the additional subband signals (copied subband signals).
• available baseband subbands may be copied and employed as highest subbands such that all subbands are filled. The same baseband subband may be copied twice or a plurality of times such that all missing subbands can be filled with values.
• Fig. 3 illustrates an upsampling process conducted by an apparatus according to an embodiment.
• a time domain audio signal 310 and some samples 315 of the audio signal 310 are illustrated.
• the audio signal is transformed in a frequency domain, e.g. a time- frequency domain to obtain a frequency-domain audio signal 320 comprising three subband signals 330.
• the analysis filter bank comprises 3 channels.
• the subband signals of the frequency domain audio signal 330 may then be replicated to obtain three additional subband signals 335 such that the frequency domain audio signal 320 comprisies the original three subband signals 330 and the generated three additional subband signals 335.
• two further additional subband signals 338 are generated, e.g.
• Fig. 4 illustrates an apparatus according to a further embodiment.
• the apparatus comprises a signal processor 405 and a configurator 408.
• the signal processor 405 comprises a core decoder module 210, an analysis filter bank 220, a subband generator 230 and a synthesis filter bank 240, which correspond to the respective units in the embodiment of Fig. 2.
• the signal processor 405 furthermore comprises an MPEG Surround decoder 410 (MPS decoder) for decoding the preprocessed audio signal to obtain a preprocessed audio signal with stereo or surround channels.
• MPS decoder MPEG Surround decoder
• the subband generator 230 is adapted to feed the frequency-domain preprocessed audio signal into the MPEG Surround decoder 410 after the additional subband signals for the frequency-domain preprocessed audio signal have been created and added to the frequency-domain preprocessed audio signal.
• Fig. 5a illustrates a core decoder module according to an embodiment.
• the core decoder module comprises a first core decoder 510 and a second core decoder 520.
• the first core decoder 510 is adapted to operate in a time domain and wherein the second core decoder 520 is adapted to operate in a frequency domain.
• the first core decoder 510 is an AC CLP decoder and the second core decoder 520 is an FD transfom decoder, e.g. an AAC transform decoder.
• the second core decoder 520 is a TCX transform decoder.
• the arriving audio signal portion asp is either processed by the ACELP decoder 510 or by the FD transform decoder 520.
• the output of the core decoder module is a prepvocessed portion of the audio signal pp-asp.
• Fig. 5b illustrates an apparatus for processing an audio signal according to the embodiment of Fig. 4 with a core decoder module according to Fig. 5 a.
• the super- frame size for the ACELP codec is reduced from 1024 to 768 samples. This could be done by combining 4 ACELP frames of size 192 (3 sub-frames of size 64) to one core-coder frame of size 768 (previously: 4 ACELP frames of size 256 were combined to a core-coder frame of size 1024).
• Fig. 6a illustrates an ACELP super frame 605 comprising 4 ACELP frames 610. Each one of the ACELP frames 610 comprises 3 sub-frames 615.
• FIG. 6b illustrates an ACELP super frame 625 comprising 3 ACELP frames 630. Each one of the ACELP frames 630 comprises 4 sub-frames 635.
• Fig. 7b outlines the proposed additional setting from a decoder perspective and compares it to the traditional USAC setting.
• Fig. 7a and 7b outline the decoder stractiire as typically used at operating points as 24 kbit/s or 32 kbit/s.
• an audio signal frame is inputted a QMF analysis filter bank 710.
• the QMF analysis filter bank 710 has 32 channels.
• the QMF analysis filter bank 710 is adapted to transform a time domain audio signal into a frequency domain, wherein the frequency domain audio signal comprises 32 subbands.
• the frequency domain audio signal is then inputted into an upsampler 720.
• the upsampler 720 is adapted to upsample the frequency domain audio signal by an upsampling factor 2.
• a frequency domain upsampler output signal comprising 64 subbands is generated by the upsampler.
• the upsampler 720 is an SBR (Spectral Band Replication) upsampler.
• Spectral Band Replication is employed to generate higher frequency subbands from lower frequency subbands being inputted into the spectral band replicator.
• the upsampled frequency domain audio signal is then fed into an MPEG Surround (MPS) decoder 730.
• MPS decoder 730 is adapted to decode a downmixed surround signal to derive frequency domain channels of a surround signal.
• the MPS decoder 730 may be adapted to generate 2 upmixed frequency domain surround channels of a frequency domain surround signal.
• the MPS decoder 730 may be adapted to generate 5 upmixcd frequency domain surround channels of a frequency domain surround signal.
• the channels of the frequency domain surround signal are then fed into the QMF synthesis filter bank 740.
• the QMF synthesis filter bank 740 is adapted to transform the channels of the frequency domain surround signal into a time domain to obtain time domain channels of the surround signal.
• the US AC decoder operates in its default setting as a 2: 1 system.
• the core- codec operates in the granularity of 1024 samples/frame at half of output sampling rate f out .
• the upsampling by a factor of 2 is implicitly performed inside the SBR tool, by combining a 32 band analysis QMF filter bank with a 64 band synthesis QMF bank running at the same rate.
• the SBR tool outputs frames of size 2048 at f 0l chorus.
• Fig. 7b illustrates the proposed extra setting for USAC.
• An QMF analysis filter bank 750, an upsampler 760, an MPS decoder 770 and a synthesis filter bank 780 are illustrated.
• the USAC codec operates in the proposed extra setting as an 8/3 system.
• the core-coder mns at 3/8 lh of the output sampling rate f beau ut .
• the core-coder frame size was scaled down by a factor of 3 ⁇ 4.
• an AAC coder employed as core coder may still determine scalefactors based on an 1 ⁇ 2 f out sampling rate, even if the AAC coder operates at 3/8 th of the output sampling rate ⁇ ⁇ ⁇
• the table below provides detailed numbers on sampling rates and frame duration for the USAC as used in the USAC reference quality encoder.
• the frame duration in the proposed new setting can be reduced by nearly 25%, which leads to positive effects for all non-stationary signals, since the spreading of coding noise can also be reduced by the same ratio. This reduction can be achieved without increasing the core-coder sampling frequency, which would have moved the ACELP tool out of its optimized operation range.
• the table illustrates sampling rates and frame duration for default and proposed new setting as used in the reference quality encoder at 24 kbit's.
• the shorter frame sizes can be easily achieved by scaling the transform and window sizes by a factor of 3 ⁇ 4.
• the FD coder in the standard mode operates with transform sizes of 1024 and 128, additional transforms of size 768 and 96 are introduced by the new setting.
• additional transfomis of size of 768, 384 and 192 are needed.
• the transform coder can remain unchanged.
• the total frame size needs to be adapted to 768 samples.
• One way to achieve this goal is to leave the overall structure of the frame is unchanged with 4 ACELP frames of 192 samples fitting within each frame of 768 samples.
• the adaptation to the reduced frame size is achieved by decreasing the number of subframes per frame from 4 to 3.
• the ACELP subframe length is unchanged at 64 samples.
• the pitch information is encoded using a slightly different scheme: three pitch values are encoded using an absolute-relative-relative scheme using 9, 6 and 6 bits respectively instead of an absolute-relative-absolute-relative scheme using 9, 6, 9 and 6 bits in the standard model.
• the other elements of the ACELP codec such as the ACELP codebooks as well as the various quantizers (I PC filters, gains, etc.), are left unchanged.
• Another way of achieving a total frame size of 768 samples would be to combine three ACELP frames of size 256 for one core-coder frame of size 768.
• the functionality of the SBR tool remains unchanged.
• the additional to the 32 band analysis band QMF, a 24 band analysis QMF is needed, to allow for an upsampling of factor 8/3.
• the impact of the proposed extra operating point on the computational complexity is explained. This is at first done on a per codec-tool base and summarized at the end. The complexity is compared against the default low sampling rate mode and against a higher sampling rate mode, as used by the USAC reference quality encoder at higher bitrates which is comparable to the corresponding HE-AACV2 setting for these operating points.
• the complexity of the transform coder parts scales with sampling rate and transform length.
• the proposed core-coder sampling rates stay roughly the same.
• the transform sizes are reduced by a factor of 3 ⁇ 4.
• the computational complexity is reduced by nearly the same factor, assuming a mixed radix approach for the underlying FFTs.
• the complexity of the transform based decoder is expected to be slightly reduced compared to the cuixent USAC operating point and reduced by a factor of 3 ⁇ 4 compared against a high-sampling operating mode.
• the complexity of the ACELP tools mainly assembles of the following operations:
• Decoding of the excitation the complexity of that operation is proportional to the number of siibframes per second, which in turn is directly proportional to the core-coder sampling frequency (the subframe size being unchanged at 64 samples). It is therefore nearly the same with the new setting.
• the expected complexity of the ACELP decoder is expected to be unchanged compared to the current USAC operating point and reduced by a factor of 3 ⁇ 4 compared against a high-sampling operating mode.
• the main contributors to the SBR complexity are the QMF filterbanks.
• the complexity here scales with sampling rate and transform size.
• the complexity of the analysis tiltevbank is reduced by roughly a factor of 3 ⁇ 4.
• the complexity of the MPEG Surround part scales with the sampling rate.
• the proposed extra operation mode has no direct impact on the complexity of the MPEG Surround tool.
• USAC RM9 operated at 34.15kHz: approx. 4.6 WMOPS;
• USAC RM9 operated at 44.1 kHz: approx. 5.6 WMOPS;
• the proposed extra operating mode requires the storage of additional MDCT window prototypes, which sum up in total to below 900 words (32 bit) additional ROM demand. In light of the total decoder ROM demand, which is roughly 25 kWord, this seems to be negligible.
• Listening test results show a significant improvement, for music and mixed test items, without degrading the quality for speech items.
• This extra setting is intended as an additional operating mode of the USAC codec.
• a listening test according to MUSHRA methodology was conducted to evaluate the performance of the proposed new setting at 24 kbit/s mono. The following conditions were contained in the test: Hidden reference; 3.5 kHz low-pass anchor; USAC WD7 reference quality (WD7@34.15kHz); USAC WD7 operated at high sampling rate (WD7@44. lkHz); and USAC WD7 reference quality, proposed new setting (WD7_CE@44.lkH2).
• the test covered the 12 test items from the USAC test set, and the following additional items: si02: castanets; velvet: electronic music; and xylophone: music box.
• Fig. 8a and 8b illustrate the results of the test. 22 subjects participated in the listening test. A Student-t probability distribution was used for the evaluation.
• WD7 operated at 44,1 kHz performs worse than WD7 for 6 items (esOl , louis_raquin, tel , WeddingSpeech, HarryPotter, SpcechOverMusic_4) and averaged over all items.
• the items it performs worse for include all pure speech items and two of (he mixed speech/music items.
• WD7 operated at 44.1 kHz performs significantly better than WD7 for four items (twinkle, salvation, si()2, velvet). All of these items contain significant portions of music signals or are classified as music.
• a new setting for mid US AC bitrates is provided.
• This new setting enables the US AC codec to increase its temporal granularity for all relevant tools, such as transform coders, SBR and MPEG Suiround, without sacrificing the quality of the ACELP tool.
• the quality for the mid bitrate range can be improved, in particular for music and mixed signals exhibiting a high temporal structure.
• the USAC systems gains at flexibility, since the USAC codec including the ACELP tool can now be used at a wider range of sampling rates, such as 44.1 kHz.
• Fig. 9 illustrates an apparatus for processing an audio signal.
• the apparatus comprises a signal processor 910 and a configurator 920.
• the signal processor 910 is adapted to receive a first audio signal frame 940 having a first configurable number of samples 945 of the audio signal. Moreover, the signal processor 910 is adapted to downsampie the audio signal by a configurable downsampling factor to obtain a processed audio signal. Furthermore, the signal processor is adapted to output a second audio signal frame 950 having a second configurable number of samples 955 of the processed audio signal.
• the configurator 920 is adapted to configure the signal processor 910 based on configuration information ci2 such that the configurable downsampling factor is equal to a first downsampling value when a first ratio of the second configurable number of samples to the first configurable number of samples has a first ratio value.
• the configurator 920 is adapted to configure the signal processor 910 such that the configurable downsampling factor is equal to a different second downsampling value, when a different second ratio of the second configurable number of samples to the first configurable number of samples has a different second ratio value.
• the first or the second ratio value is not an integer value.
• An apparatus according to Fig. 9 may for example be employed in the process of encoding.
• aspects have been described in the context of an apparatus, it is clear that these aspects also represent a description of the corresponding method, where a block or device corresponds to a method step or a feature of a method step. Analogously, aspects described in the context of a method step also represent a description of a corresponding block or item or feature of a corresponding apparatus.
• the inventive decomposed signal can be stored on a digital storage medium or can be transmitted on a transmission medium such as a wireless transmission medium or a wired transmission medium such as the Internet.
• embodiments of the invention can be implemented in hardware or in software.
• the implementation can be performed using a digital storage medium, for example a floppy disk, a DVD, a CD, a ROM, a PROM, an EPROM, an EEPROM or a FLASH memory, having electronically readable control signals stored thereon, which cooperate (or are capable of cooperating) with a programmable computer system such that the respective method is performed.
• Some embodiments according to the invention comprise a non-transitory data carrier having electronically readable control signals, which are capable of cooperating with a programmable computer system, such that one of the methods described herein is performed.
• embodiments of the present invention can be implemented as a computer program product with a program code, the program code being operative for perfomiing one of the methods when the computer program product inns on a computer.
• the program code may for example be stored on a machine readable carrier.
• Other embodiments comprise the computer program for performing one of the methods described herein, stored on a machine readable carrier.
• an embodiment of the inventive method is, therefore, a computer program having a program code for performing one of the methods described herein, when the computer program runs on a computer.
• a further embodiment of the inventive methods is, therefore, a data carrier (or a digital
• a further embodiment of the inventive method is, therefore, a data stream or a sequence of signals representing the computer program for performing one of the methods described .5 herein.
• the data stream or the sequence of signals may for example be configured to be transferred via a data communication connection, for example via the Internet.
• a further embodiment comprises a processing means, for example a computer, or a programmable logic device, configured to or adapted to perform one of the methods 20 described herein.
• a processing means for example a computer, or a programmable logic device, configured to or adapted to perform one of the methods 20 described herein.
• a further embodiment comprises a computer having installed thereon the computer program for performing one of the methods described herein.
• a programmable logic device for example a field programmable gate array
• a field programmable gate array may cooperate with a microprocessor in order to perform one of the methods described herein.
• the methods are preferably performed by any hardware apparatus.

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