WO2016005690A1 - Mise a jour des états d'un post-traitement a une fréquence d'échantillonnage variable selon la trame - Google Patents
Mise a jour des états d'un post-traitement a une fréquence d'échantillonnage variable selon la trame Download PDFInfo
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- WO2016005690A1 WO2016005690A1 PCT/FR2015/051864 FR2015051864W WO2016005690A1 WO 2016005690 A1 WO2016005690 A1 WO 2016005690A1 FR 2015051864 W FR2015051864 W FR 2015051864W WO 2016005690 A1 WO2016005690 A1 WO 2016005690A1
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- 238000005070 sampling Methods 0.000 title claims abstract description 60
- 238000012805 post-processing Methods 0.000 title claims abstract description 48
- 238000000034 method Methods 0.000 claims abstract description 44
- 230000015654 memory Effects 0.000 claims abstract description 39
- 238000012545 processing Methods 0.000 claims abstract description 18
- 239000004149 tartrazine Substances 0.000 claims abstract description 4
- 239000004229 Alkannin Substances 0.000 claims abstract description 3
- 239000002151 riboflavin Substances 0.000 claims abstract description 3
- 238000012952 Resampling Methods 0.000 claims description 24
- 238000004590 computer program Methods 0.000 claims description 7
- 230000006870 function Effects 0.000 claims description 2
- 230000005236 sound signal Effects 0.000 abstract description 2
- 238000001914 filtration Methods 0.000 description 12
- 230000008569 process Effects 0.000 description 8
- 230000008859 change Effects 0.000 description 5
- 239000007787 solid Substances 0.000 description 4
- 230000005284 excitation Effects 0.000 description 3
- 230000007704 transition Effects 0.000 description 3
- 230000003044 adaptive effect Effects 0.000 description 2
- 230000009467 reduction Effects 0.000 description 2
- 230000005540 biological transmission Effects 0.000 description 1
- 230000015572 biosynthetic process Effects 0.000 description 1
- 238000004891 communication Methods 0.000 description 1
- 238000010586 diagram Methods 0.000 description 1
- 239000003623 enhancer Substances 0.000 description 1
- 230000006872 improvement Effects 0.000 description 1
- 238000004519 manufacturing process Methods 0.000 description 1
- 230000005055 memory storage Effects 0.000 description 1
- 238000011084 recovery Methods 0.000 description 1
- 238000003786 synthesis reaction Methods 0.000 description 1
Classifications
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- 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
- G10L19/16—Vocoder architecture
- G10L19/18—Vocoders using multiple modes
- G10L19/24—Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding
-
- 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
- G10L19/26—Pre-filtering or post-filtering
-
- 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
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- 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
- G10L19/16—Vocoder architecture
- G10L19/18—Vocoders using multiple modes
Definitions
- the present invention relates to the processing of an audiofrequency signal for its transmission or storage. More particularly, the invention relates to an update of the states of a post-processing of a decoded audio-frequency signal, when the sampling frequency varies from one signal frame to another.
- the invention applies more particularly to the case of linear prediction decoding such as CELP ("Coded Excitation Linear Prediction") type decoding.
- Linear prediction coded algorithms such as "Algebraic Coded Excitation” (ACELP) type codecs, are known to be suitable for speech signals and are well modeled for their production.
- ACELP Algebraic Coded Excitation
- sampling frequency at which the CELP coding algorithm operates is generally predetermined and identical in each coded frame; examples of sampling frequencies are:
- s p (n) 0.5s (n - T) + 0.5i (/ z + T)
- This processing requires a memory of the past signal whose size must cover the different possible pitch values T (to know the value s (n - T)).
- the value of the pitch T is not known for the next frame, so in general, to cover the worst possible case, MAXPITCH + 1 samples of the decoded signal passed are memorized for the postprocessing.
- MAXPITCH gives the maximum pitch length at the given sampling frequency, for example in general this value is 289 to 16 kHz or 231 to 12.8 kHz.
- An additional sample is often stored and then performs a de-emphasis filtering of order 1. We will not detail here this deemphasis filtering which is not the subject of the present invention.
- ITU-T G.722.2 the wideband input and output signal is sampled at 16 kHz but the CELP coding operates at 12.8 kHz. Note that the ITU-T G.718 and G.718 Annex C codes also work with 8 and / or 32 kHz input / output frequencies, with a 12.8 kHz CELP core.
- the input signal is normally broadband (at 16 kHz) and the low band (0-4 kHz) is obtained by a QMF filter bank to obtain a signal sampled at 8 kHz before coding by a CELP algorithm derived from ITU-T G.729 and G.729 Annex A.
- fs 1 and fs 2 16kHz.
- a change in rate over time, from one frame to another may in this case cause switching between these two frequencies (fs 1 and fs 2 ) depending on the range of flow rates covered. This frequency switching between two frames can cause audible and annoying artifacts for several reasons.
- one option is to disable post-processing over the duration of the transition frame (the frame after the internal sampling rate change). This option does not produce a desirable result in general because the noise that was post-filtered reappears abruptly on the transition frame.
- Another option is to leave the post-processing active but by setting the memories to zero. With this method, the quality obtained is very poor.
- Another possibility is also to consider a memory at 16 kHz as if it were at 12.8 kHz by keeping only the last 4/5 samples of this memory or conversely, to consider a memory at 12.8 kHz as if it were at 16 kHz, either by adding 1/5 of zeros at the beginning (towards the past) of this memory to have the right length, either by memorizing 20% more samples at 12.8 kHz to have enough in case of change of internal sampling frequency . The listening tests show that these solutions do not give a satisfactory quality.
- the present invention improves the situation.
- the method proposes a method for updating the states of a post-processing applied to a decoded audio-frequency signal.
- the method is such that, for a current decoded signal frame, sampled at a sampling frequency different from the previous frame, it comprises the following steps:
- the post-processing memory is adapted to the sampling frequency of the current frame that is post-processed. This technique improves post-processing quality in the transition frames between two sampling rates while minimizing the increase in complexity (computational load, ROM, RAM and PROM).
- the interpolation is performed starting from the most recent sample of the decoded signal. passed and interpolating in the reverse chronological direction and in the case where the sampling frequency of the previous frame is lower than the sampling frequency of the current frame, the interpolation is performed starting from the oldest sample decoded signal passed and interpolating in the chronological direction.
- This interpolation mode makes it possible to use only one storage array (of length corresponding to the maximum of the signal period for the largest sampling frequency) to record the decoded signal passed before and after resampling. Indeed, in both directions of resampling, the interpolation is adapted to the fact that from the moment when a sample of the past signal is used for an interpolation, it is no longer used for the following interpolation. It can thus be replaced by the interpolated one in the storage array.
- the decoded signal passed, resampled is stored in the same buffer as the decoded signal passed before resampling.
- the interpolation is of linear type.
- the decoded signal passed is of fixed length depending on a possible maximum of speech signal period.
- the state update method is particularly suitable in the case where the postprocessing is applied to the decoded signal on a low frequency band to decrease the low frequency noise.
- the invention also relates to a method for decoding a current frame of an audiofrequency signal comprising a step of selecting a decoding sampling frequency, a post-processing step.
- the process is such that in the where the preceding frame is sampled at a first sampling frequency different from a second sampling frequency of the current frame, it comprises updating the states of the post-processing according to a method as described.
- the low-frequency processing of the decoded signal is therefore adapted to the internal sampling frequency of the decoder, the quality of this post-processing being then improved.
- the invention relates to a device for processing a decoded audio-frequency signal, characterized in that it comprises, for a current frame of decoded signal, sampled at a sampling frequency different from the previous frame:
- a resampling module for resampling at the sampling frequency of the current frame, by interpolation, the obtained decoded signal obtained
- a post-processing module using the decoded signal passed resampled as a memory of the post-processing of the current frame.
- This device has the same advantages as the method described above, which it implements.
- the present invention also relates to an audio-frequency signal decoder comprising a module for selecting a decoding sampling frequency and at least one processing device as described.
- the invention relates to a computer program comprising code instructions for implementing the steps of the method of updating the states as described, when these instructions are executed by a processor.
- the invention relates to a storage medium, readable by a processor, integrated or not to the processing device, possibly removable, storing a computer program implementing a method of updating the states as described above.
- FIG. 1 illustrates in flowchart form a method for updating the states of a post-processing according to one embodiment of the invention
- FIG. 2 illustrates an example of resampling from 16 kHz to 12.8 kHz, according to one embodiment of the invention
- FIG. 3 illustrates an example of resampling of, 12.8kHz to 16kHz according to one embodiment of the invention
- FIG. 4 illustrates an exemplary decoder comprising decoding modules operating at different sampling frequencies, and a processing device according to one embodiment of the invention
- FIG. 5 illustrates a hardware representation of a processing device according to one embodiment of the invention.
- FIG. 1 illustrates in flowchart form the steps implemented in the process of updating the states of a post-processing according to one embodiment of the invention.
- the method according to one embodiment of the invention applies when the internal decoding frequency CELP in the current frame (/ 3 ⁇ 4) is different from the internal frequency of CELP decoding of the previous frame (/ 3 ⁇ 4): fsi ⁇ fs 2
- the CELP coder or decoder has two internal sampling frequencies: 12.8kHz for low bit rates and 16kHz for high bit rates. Of course, other internal sampling frequencies may be provided within the scope of the invention.
- the process for updating the post-processing states implemented on a decoded audio-frequency signal comprises a first step E101 of recovery in a buffer buffer, also called a buffer, of a decoded signal passed, memorized during the decoding of the previous frame .
- this decoded signal of the previous frame (Mem fsi) is at a first internal sampling frequency fsi.
- the memorized decoded signal length is a function, for example, of the maximum value of the speech signal period (or "pitch").
- the maximum value of the coded pitch is 289.
- the same memory buffer of 290 samples is used for both cases, at 16 kHz all the indices from 0 to 289 are necessary, at 12.8 kHz only the indices of 58 to 289 are useful.
- the last sample of the memory (index 289) therefore always contains the last sample of the decoded signal passed, regardless of the sampling frequency. Note that at the two sampling frequencies (12.8 kHz or 16 kHz) the memory covers the same time support, 18.125 ms. Note also that at 12.8 kHz it is also possible to use the indices from 0 to 231 and to ignore the samples from 232 to 289. Intermediate positions are also possible but from a programming point of view these solutions are not not practical. In the preferred implementation of the invention the first solution is used (indices 58 to 289).
- this decoded past signal is resampled at the internal sampling frequency of the current frame 3 ⁇ 4.
- This resampling is performed for example by a linear interpolation method of low complexity.
- Other types of interpolations may be used such as cubic interpolation or "splines" for example.
- the interpolation used makes it possible to use only one RAM storage array (a single buffer memory).
- the figure also illustrates how these signals are stored in the buffer.
- the samples stored at 12.8 kHz are aligned with the end of the mem buffer (whichever is preferred).
- the numbers give the index of the location in the storage array. Empty round dotted markers of index 0 to 3 correspond to unused locations at 12.8 kHz.
- pf5 is a table pointer (addressing) for the 16 kHz input signal
- pf4 is a table pointer for the 12.8 kHz output signal.
- nb_block contains the number of blocks to process in the for loop.
- pf4 [0] is the array value pointed to by the pointer pf4
- pf4 [-l] is the previous value and so on. It's the same for pf5.
- the pointers pf5 and pf4 recede in steps of 5 and 4 samples respectively.
- Part b) of Figure 2 illustrates the case where samples at 12.8 kHz are aligned with the beginning of the buffer "mem" and the locations of the index 16 to 19 are not used. In this case, as illustrated by the full arrow, interpolation must start from the oldest sample to be able to rewrite the result in the same table.
- the empty square at 16 kHz represents the beginning of the decoded signal of the current frame. Note that the first sample of the current frame at 16 kHz is identical to that at 12.8 kHz (same time time), this is represented by an empty circle.
- the dotted arrows give for each 16 kHz output sample the 12.8 kHz input samples from which they are interpolated in the case of linear interpolation. For the interpolation of the last output sample one must also use the first sample of the current frame at 12.8 kHz, which is well known as explained above. This dependence is illustrated by a broken arrow in FIG.
- the figure also shows how these signals are stored in the buffer, the numbers give the index of the location in the array.
- the samples stored at 12.8 kHz are aligned with the end of the buffer "mem" (according to the preferred implementation). Empty round dotted markers from index 0 to 3 correspond to locations not available (because not used) at 12.8 kHz
- the interpolation is carried out starting from the oldest sample (thus that of index 0 at the output) to be able to rewrite the result of the interpolation in the same table of memory because the old value in these locations is not used to perform the following interpolations.
- the solid arrow shows the direction of the interpolation, the numbers written in the arrow correspond to the order in which the output samples are interpolated.
- pf4 is a table pointer for the input signal at 12.8 kHz which points to the beginning of the filter memory, this memory is stored from the nb_bloâ me sample of the tableau mem.
- pf5 is a table pointer for the 16 kHz output signal, it points to the first element of the mem array.
- nb_block contains the number of blocks to process.
- nb_block-l blocks are processed in the for loop, and the last block is processed separately.
- pf4 [0] is the array value pointed to by the pointer pf4,
- pf4 [l] is the next value and so on. It's the same for pf5.
- the pointers pf5 and pf4 advance in steps of 5 and 4 samples respectively.
- the decoded signal of the current frame is stored in the syn array, syn [0] is the first sample of the current frame
- Part b) of Figure 3 illustrates the case where the samples at 12.8 kHz are aligned with the beginning of the buffer "mem" and the locations of the index 16 to 19 are not used. In this case, as shown by the full arrow, you need to interpolate from the most recent sample to be able to rewrite the result in the same table.
- step E102 of resampling of the memory Mem. FSI. at the frequency / 3 ⁇ 4 we obtain the memory or decoded signal passed, resampled (Mem fs2). This decoded signal passed resampled is used in step E103 as a new postprocessing memory of the current frame.
- the post-processing is similar to that described in ITU-T Recommendation G.718.
- FIG. 4 now describes an exemplary decoder comprising a processing device 410 in one embodiment of the invention.
- the output signal y (n) (mono) is sampled at the frequency fs out which can take the values of 8, 16, 32 or 48 kHz.
- the bitstream is de-multiplexed at 401 and decoded.
- the decoder determines at 402, here according to the rate of the current frame, how often fs 1 or fs 2 decode the information from a CELP coder.
- either the decoding module 403 for the frequency fs 1 or the decoding module 404 for the frequency 3 ⁇ 4 is implemented to decode the received signal.
- CELP decoding at 16 kHz is not detailed here because it goes beyond the scope of the invention.
- the output of the CELP decoder in the current frame is then post-filtered by the processing device 410 implementing the process of updating the post-processing states described with reference to FIG. 1.
- This device comprises post-processing modules. 420 and 421 adapted to the respective sampling frequencies fsj and 3 ⁇ 4 able to carry out a low-frequency noise reduction type after-treatment, also called low-frequency post-filtering, similarly to the "bass post-filter" (BPF) of the ITU-T T G.718, using the post-processing memories resampled by the resampling module 422.
- the processing device also comprises a resampling module 422 resampling a decoded signal passed , stored for the previous frame, by interpolation.
- the decoded signal passed from the previous frame (Mem fsj), sampled at the frequency fsj, is resampled at the frequency s 2 to obtain a decoded past signal resampled (Mem 3 ⁇ 4) used as a post-memory. processing of the current frame.
- fsfi used as the post-processing memory of the current frame.
- a high band signal (resampled at the frequency fs out ) decoded by the decoding module 405 can be added at 406 to the resampled low band signal.
- the decoder also provides for the use of additional decoding modes, such as inverse frequency transform decoding (block 430) in the case where the input signal to be encoded has been coded by a transform coder.
- additional decoding modes such as inverse frequency transform decoding (block 430) in the case where the input signal to be encoded has been coded by a transform coder.
- the coder analyzes the type of signal to be coded and chooses the coding technique best suited to this signal.
- Transform coding is mainly used for musical signals that are generally badly coded by a CELP-type predictive coder.
- FIG. 5 shows an example of a hardware embodiment of a processing device 500 according to one embodiment of the invention. This may be an integral part of an audio-frequency signal decoder or equipment receiving audio-frequency signals. It can be integrated into a communication terminal, a set-top box set-top box or a home gateway.
- This type of device comprises a processor PROC 506 cooperating with a memory block BM having a memory storage and / or work MEM.
- Such a device comprises an input module 501 adapted to receive audio signal frames and in particular a memorized part (Buf pre c) of a previous frame at a first sampling frequency / ⁇ .
- It comprises an output module 502 capable of transmitting a current frame of post-processed audio frequency signal s' (n).
- the processor PROC controls the obtaining module 503 of a decoded signal passed, stored for the previous frame. Typically, the obtaining of this decoded past signal is performed by simple reading in a memory of the buffer type, included in the memory block BM.
- the processor also controls a resampling module 504 to re-sample by interpolation the past decoded signal obtained at 503.
- the memory block may advantageously comprise a computer program comprising code instructions for implementing the steps of the process for updating the post-processing states in the sense of the invention, when these instructions are executed by the processor PROC, and in particular the steps of obtaining a decoded signal passed, stored for the previous frame, of resampling by interpolation of the obtained decoded signal obtained and of using the decoded signal passed resampled as a memory of the post-processing of the current frame.
- FIG. 1 repeats the steps of an algorithm of such a computer program.
- the computer program can also be stored on a memory medium readable by a reader of the device or downloadable in the memory space thereof.
- the memory MEM generally records all the data necessary for the implementation of the method.
- nb_bloc len_mem_16 / 5;
- nb_bloc len_mem_16 / 5;
- pf5 [1] 0.2f * pf4 [0] -f 0. .8f * pf4 [1]
- pf5 [2] 0.4f * pf4 [1] -f O.fpf4 [2]
- pf5 [3] 0.6f * pf4 [2] -f 0..4f * pf4 [3]
- pf5 [4] 0.8f * pf4 [3] -f 0. .2f * syn [0]
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Abstract
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Priority Applications (6)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
US15/325,643 US10424313B2 (en) | 2014-07-11 | 2015-07-06 | Update of post-processing states with variable sampling frequency according to the frame |
CN201580037397.4A CN106489178B (zh) | 2014-07-11 | 2015-07-06 | 利用根据帧的可变采样频率对后处理状态进行更新 |
JP2017500355A JP6607915B2 (ja) | 2014-07-11 | 2015-07-06 | フレームに基づく可変サンプリング周波数による後処理状態の更新 |
KR1020177003571A KR102271224B1 (ko) | 2014-07-11 | 2015-07-06 | 프레임에 따른 가변 샘플링 주파수에 의한 후처리 상태들의 업데이트 |
EP15742373.2A EP3167447B1 (fr) | 2014-07-11 | 2015-07-06 | Mise a jour des états d'un post-traitement a une fréquence d'échantillonnage variable selon la trame |
ES15742373.2T ES2686349T3 (es) | 2014-07-11 | 2015-07-06 | Actualización de los estados de un postratamiento a una frecuencia de muestreo variable según la trama |
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FR1456734A FR3023646A1 (fr) | 2014-07-11 | 2014-07-11 | Mise a jour des etats d'un post-traitement a une frequence d'echantillonnage variable selon la trame |
FR1456734 | 2014-07-11 |
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US (1) | US10424313B2 (fr) |
EP (1) | EP3167447B1 (fr) |
JP (1) | JP6607915B2 (fr) |
KR (1) | KR102271224B1 (fr) |
CN (1) | CN106489178B (fr) |
ES (1) | ES2686349T3 (fr) |
FR (1) | FR3023646A1 (fr) |
WO (1) | WO2016005690A1 (fr) |
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FR3023646A1 (fr) * | 2014-07-11 | 2016-01-15 | Orange | Mise a jour des etats d'un post-traitement a une frequence d'echantillonnage variable selon la trame |
EP2988300A1 (fr) * | 2014-08-18 | 2016-02-24 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Commutation de fréquences d'échantillonnage au niveau des dispositifs de traitement audio |
CN111223491B (zh) * | 2020-01-22 | 2022-11-15 | 深圳市倍轻松科技股份有限公司 | 一种提取音乐信号主旋律的方法、装置及终端设备 |
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CN109448745B (zh) * | 2013-01-07 | 2021-09-07 | 中兴通讯股份有限公司 | 一种编码模式切换方法和装置、解码模式切换方法和装置 |
FR3001593A1 (fr) * | 2013-01-31 | 2014-08-01 | France Telecom | Correction perfectionnee de perte de trame au decodage d'un signal. |
FR3015754A1 (fr) * | 2013-12-20 | 2015-06-26 | Orange | Re-echantillonnage d'un signal audio cadence a une frequence d'echantillonnage variable selon la trame |
FR3023646A1 (fr) * | 2014-07-11 | 2016-01-15 | Orange | Mise a jour des etats d'un post-traitement a une frequence d'echantillonnage variable selon la trame |
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2014
- 2014-07-11 FR FR1456734A patent/FR3023646A1/fr not_active Withdrawn
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2015
- 2015-07-06 CN CN201580037397.4A patent/CN106489178B/zh active Active
- 2015-07-06 WO PCT/FR2015/051864 patent/WO2016005690A1/fr active Application Filing
- 2015-07-06 KR KR1020177003571A patent/KR102271224B1/ko active IP Right Grant
- 2015-07-06 EP EP15742373.2A patent/EP3167447B1/fr active Active
- 2015-07-06 US US15/325,643 patent/US10424313B2/en active Active
- 2015-07-06 ES ES15742373.2T patent/ES2686349T3/es active Active
- 2015-07-06 JP JP2017500355A patent/JP6607915B2/ja active Active
Non-Patent Citations (4)
Title |
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"ISO/IEC 14496-3:2001(E) - Subpart 3: Speech Coding - CELP", INTERNATIONAL STANDARD ISO/IEC, XX, XX, 1 January 2001 (2001-01-01), pages 1 - 172, XP007902532 * |
"ITU-T G.718 - Frame error robust narrow-band and wideband embedded variable bit-rate coding of speech and audio from 8-32 kbit/s", 30 June 2008 (2008-06-30), XP055087883, Retrieved from the Internet <URL:http://www.itu.int/rec/T-REC-G.718-200806-I> [retrieved on 20131112] * |
C. LAFLAMME ET AL.: "wideband speech coding technique based on algebraic CELP", ICASSP, 1991 |
G. ROY; P. KABAL: "Wideband CELP speech coding at 16 kbits/sec", ICASSP, 1991 |
Also Published As
Publication number | Publication date |
---|---|
CN106489178B (zh) | 2019-05-07 |
CN106489178A (zh) | 2017-03-08 |
US20170148461A1 (en) | 2017-05-25 |
FR3023646A1 (fr) | 2016-01-15 |
EP3167447B1 (fr) | 2018-06-06 |
KR102271224B1 (ko) | 2021-06-29 |
EP3167447A1 (fr) | 2017-05-17 |
JP2017521714A (ja) | 2017-08-03 |
JP6607915B2 (ja) | 2019-11-20 |
KR20170028988A (ko) | 2017-03-14 |
US10424313B2 (en) | 2019-09-24 |
ES2686349T3 (es) | 2018-10-17 |
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