WO2002063609A1 - Methode et dispositif de traitement d'une pluralite de flux binaires audio - Google Patents
Methode et dispositif de traitement d'une pluralite de flux binaires audio Download PDFInfo
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- WO2002063609A1 WO2002063609A1 PCT/FR2002/000366 FR0200366W WO02063609A1 WO 2002063609 A1 WO2002063609 A1 WO 2002063609A1 FR 0200366 W FR0200366 W FR 0200366W WO 02063609 A1 WO02063609 A1 WO 02063609A1
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- WO
- WIPO (PCT)
- Prior art keywords
- audio
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- Prior art date
Links
- 238000012545 processing Methods 0.000 title claims abstract description 20
- 238000000034 method Methods 0.000 title claims description 18
- 230000005236 sound signal Effects 0.000 claims abstract description 42
- 238000011069 regeneration method Methods 0.000 claims description 54
- 230000008929 regeneration Effects 0.000 claims description 52
- 230000000873 masking effect Effects 0.000 claims description 42
- 230000005540 biological transmission Effects 0.000 claims description 31
- 230000003595 spectral effect Effects 0.000 claims description 10
- 238000003672 processing method Methods 0.000 claims description 8
- 230000009466 transformation Effects 0.000 claims description 5
- 238000001228 spectrum Methods 0.000 claims description 4
- 230000006870 function Effects 0.000 description 9
- 230000000694 effects Effects 0.000 description 6
- 230000004907 flux Effects 0.000 description 5
- 230000003111 delayed effect Effects 0.000 description 4
- 238000012937 correction Methods 0.000 description 3
- 230000015654 memory Effects 0.000 description 3
- 238000013139 quantization Methods 0.000 description 3
- 238000012360 testing method Methods 0.000 description 3
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Classifications
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L1/00—Arrangements for detecting or preventing errors in the information received
- H04L1/20—Arrangements for detecting or preventing errors in the information received using signal quality detector
- H04L1/208—Arrangements for detecting or preventing errors in the information received using signal quality detector involving signal re-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
- G10L19/005—Correction of errors induced by the transmission channel, if related to the coding algorithm
-
- 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
- G10L19/012—Comfort noise or silence coding
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L1/00—Arrangements for detecting or preventing errors in the information received
- H04L1/20—Arrangements for detecting or preventing errors in the information received using signal quality detector
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M3/00—Automatic or semi-automatic exchanges
- H04M3/42—Systems providing special services or facilities to subscribers
- H04M3/56—Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M3/00—Automatic or semi-automatic exchanges
- H04M3/42—Systems providing special services or facilities to subscribers
- H04M3/56—Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities
- H04M3/568—Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities audio processing specific to telephonic conferencing, e.g. spatial distribution, mixing of participants
Definitions
- a discontinuity in an audio stream can be due to the loss of part of the audio stream data, to jitter and / or asynchronism between the receiver and the transmitter or to a discontinuous transmission of the transmitter.
- the loss of part of the audio stream generally corresponds to the loss of one or more frames, the length of the frames being dependent on the coding and the transmission mode specific to each network.
- the quality of service is generally not guaranteed.
- the network has no minimum threshold on the bit rate offered for packet transport. The bandwidth available depends on the traffic conditions on the network. Thus, in the case of heavy congestion, the fluidity of transmission is no longer possible. An overflow of the routers' buffers can therefore occur and certain packets can be rejected.
- Some audio transmitters incorporate a silence compression system associated with a discontinuous transmission mechanism to reduce the bit rate transmitted during the silence intervals and thus preserve the bandwidth allocated to the communication.
- VAD Voice Activity Detector / Discontinuous Transmission / Comfort Noise Generator
- a voice activity detector VAD
- VAD Voice Activity detector
- a discontinuous transmission module decides either to extend the voice activity ("hangover") to avoid a too sudden cut at the end of an active period, or to send or not a frame SID (Silence Insertion Descriptor).
- This SID frame contains coded information describing the characteristics of the background noise to allow the insertion of artificial noise to the receiver during periods of inactivity.
- the background noise characteristics may or may not be updated during the period of voice inactivity.
- the update frequency also depends on the DTX discontinuous transmission system: some DTX send an SID frame at regular intervals (GSM), others only send an SID frame when the power spectrum noise has changed (ITU-T G.723.1).
- GSM regular intervals
- ITU-T G.723.1 When the system does not update the characteristics of the background noise, the information for the end of the voice activity period may or may not be indicated in the last frame transmitted.
- some audio transmitters, operating in DTX mode do not transmit an SID frame or provide information about the end of the voice activity period.
- the implementation of the regeneration techniques described above increases the complexity of the audio processing devices and more particularly of the multi-stream processing devices.
- the systematic regeneration of all the non-received streams not only leads to an increase in complexity, but it also risks deteriorating the quality if the coder is poorly performing for multi-source signals or if it resists transcoding cascades poorly.
- the coding used is of the CELP (Code Excited Linear Predictive) type at low bit rate.
- the multipoint control unit also called audio bridge, performs the combination (or mixing) of a plurality of audio streams then routing to the terminals concerned.
- Fig. 1 illustrates such an audio bridge.
- the bridge From K input audio streams received Ei, .. E ⁇ , the bridge supplies K output audio streams S ⁇ , .., S ⁇ by combination of the input streams in the module (130).
- the sum of the (Kl) input flows (E ⁇ ), p ⁇ m is directed to the output Sm.
- the input streams Prior to their combination, the input streams are decoded or partially decoded in the decoders 110 ⁇ , 110, .. 110 ⁇ - Conversely, the output streams undergo a recoding or a partial recoding in the encoders 120j, 120 2 , ... 120 ⁇ .
- the input streams E p consist of coded audio data frames.
- the coding used can be a perceptual coding effecting a modified discrete cosine transform (TCDM) on the digital frames of the audio signal followed by a quantification of the frequency components obtained with dynamic allocation of bits as a function of the masking levels by sub-bands.
- TDAC modified discrete cosine transform
- the structure of this perceptual coder is illustrated in Fig. 2.
- the unit 250 normalizes each transform coefficient y (k) by the energy e q (j) of the sub-band to which it belongs and quantifies each coefficient thus normalized as a function of the number of allocated bits.
- voicing v ⁇ in the following way: if all the input frames p, p ⁇ m are of the same nature, voiced or unvoiced, the combined voicing information, v_ "will respectively indicate voicing or non voicing. If the input frames p, p ⁇ m are of different natures, the unit 380 m calculates the sum of the energies of the voiced frames on the one hand, and the sum of the energies of the unvoiced frames on the other hand, and determines the combined voicing information, vP, in comparing these two sums. Finally, the unit 385 m determines the combined tone information W as follows: if all the input frames p, p ⁇ m have the same tone information, the combined tone information will take the common value. On the other hand, if the tone information differs, the determination of the combined information is carried over to the partial coding phase.
- the partial recoder illustrated in Fig. 3C receives from the mixing module 130 the combined values yc (k), e? (J), vF t t ⁇ from an output m of the mixing module.
- the tone information is calculated by the unit 362 from the coefficients y c (k) if it could not be determined by the mixing module 130.
- the values of the energies e (j) as well as the information of voicing v e are used by a unit 332 of spectral envelope encoding for quantizing and coding the energy of the different sub-bands.
- the values of the energies e (j) are also used by the unit 342 for calculating the masking curve. The latter provides the dynamic allocation unit 322 as well as the masking unit 352 with the masking levels for the different sub-bands j.
- the invention is also defined by an audio signal processing device adapted to receive a plurality of input audio streams and to provide at least one output audio stream from said input audio streams, at least one of the input audio stream, said first stream, which may have missing data, said device comprising means for deciding whether or not to regenerate at least part of the missing data of said first stream from at least one characteristic of at least one second input audio stream.
- the partial decoding means de-quantify the frequency components of said audio signals by sub-bands
- the regeneration decision means calculate the energy in each sub-band of each audio signal and determine, for each sub-band, the signal having the highest energy and a level of masking which it induces in said sub-band, the generation of missing data of an audio signal in said sub-band resulting from a comparison, said subsequent comparison, of the energy of the frequency components of this signal in said sub-band with said masking level.
- an audio terminal for a teleconferencing system, adapted to receive and mix a plurality of input audio streams to provide an output audio signal, the input audio streams being able to present missing data
- said terminal comprising a plurality of partial decoding means supplying partially decoded data streams from the input audio streams, regeneration decision means operating on the partially decoded data streams and deciding whether to regenerate missing data by means of regeneration, means of summing partially decoded or regenerated data and means of frequency / time transformation supplying said audio output signal from the summed values provided by said summing means, said decision means being adapted to decide the regeneration of at least part of the missing data of a first input stream from at least one characteristic of at least a second input audio stream.
- FIG. 3C schematically shows an example of a partial encoder for the audio bridge illustrated in FIG. 1;
- the general idea underlying the invention is to proceed, in the event of discontinuity of an audio stream, to the regeneration of the missing data according to a decisional criterion.
- This criterion can be in particular a perceptual criterion based on the inter-signal masking properties or a criterion depending on the complexity of the regeneration operation and the operational resources at the time of the decision.
- Each register stores the dequantized components y ° (k), the dequantized energy values' Q) as well as the voicing and tone information V and f relating to a flow / 7.
- the decision module detects the absence of data or the transmission silence and decides whether or not to regenerate the missing data. If the flow of E p is discontinuous and the module 540 has made a regeneration decision, the missing data will be totally or partially regenerated in the regeneration unit 550 p before the frame is transmitted to the mixing unit 530. On the other hand, if the flow E p is continuous, the partially decoded frame is transmitted directly to the mixing unit. In both cases, the frames from the different streams are combined in the mixing module 530 and the combined streams are recoded in the partial coders 520 ⁇ to 520 ⁇ to produce the output audio streams in S ⁇ , .., S ⁇ .
- condition (2) is not fulfilled, in other words if the signal not received is no longer masked, the band is regenerated there. Conversely, the signal not received is still masked and there is no need to regenerate.
- the module 540 determines, for each index m ⁇ p 0 of output, if at least one of the input flows /? entering into the composition of the output stream m is active, whether it is received or regenerated. If there exists for any index m ⁇ po such an active flux, the comfort noise is not generated. On the other hand, if there exists an index m ⁇ po such that none of the input streams entering into the composition of the output stream m is active, then the conditional generation procedure by strip described above is applied.
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- Engineering & Computer Science (AREA)
- Signal Processing (AREA)
- Multimedia (AREA)
- Audiology, Speech & Language Pathology (AREA)
- Computational Linguistics (AREA)
- Health & Medical Sciences (AREA)
- Computer Networks & Wireless Communication (AREA)
- Human Computer Interaction (AREA)
- Physics & Mathematics (AREA)
- Acoustics & Sound (AREA)
- Quality & Reliability (AREA)
- Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
- Compression, Expansion, Code Conversion, And Decoders (AREA)
Abstract
Description
Claims
Priority Applications (2)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
DE60224005T DE60224005T2 (de) | 2001-02-02 | 2002-01-31 | Verfahren und vorrichtung zur verarbeitung von mehreren audiobitströmen |
EP02701358A EP1356455B1 (fr) | 2001-02-02 | 2002-01-31 | Methode et dispositif de traitement d'une pluralite de flux binaires audio |
Applications Claiming Priority (2)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
FR0101662A FR2820573B1 (fr) | 2001-02-02 | 2001-02-02 | Methode et dispositif de traitement d'une pluralite de flux binaires audio |
FR01/01662 | 2001-02-02 |
Publications (1)
Publication Number | Publication Date |
---|---|
WO2002063609A1 true WO2002063609A1 (fr) | 2002-08-15 |
Family
ID=8859733
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
PCT/FR2002/000366 WO2002063609A1 (fr) | 2001-02-02 | 2002-01-31 | Methode et dispositif de traitement d'une pluralite de flux binaires audio |
Country Status (4)
Country | Link |
---|---|
EP (1) | EP1356455B1 (fr) |
DE (1) | DE60224005T2 (fr) |
FR (1) | FR2820573B1 (fr) |
WO (1) | WO2002063609A1 (fr) |
Cited By (3)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US8972033B2 (en) | 2006-10-11 | 2015-03-03 | The Nielsen Company (Us), Llc | Methods and apparatus for embedding codes in compressed audio data streams |
US9191581B2 (en) | 2004-07-02 | 2015-11-17 | The Nielsen Company (Us), Llc | Methods and apparatus for mixing compressed digital bit streams |
US9202256B2 (en) | 2003-06-13 | 2015-12-01 | The Nielsen Company (Us), Llc | Methods and apparatus for embedding watermarks |
Families Citing this family (14)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
CN1739141A (zh) * | 2003-02-06 | 2006-02-22 | 杜比实验室特许公司 | 连续的备份音频 |
AU2004248544B2 (en) | 2003-05-28 | 2010-02-18 | Dolby Laboratories Licensing Corporation | Method, apparatus and computer program for calculating and adjusting the perceived loudness of an audio signal |
AU2005299410B2 (en) | 2004-10-26 | 2011-04-07 | Dolby Laboratories Licensing Corporation | Calculating and adjusting the perceived loudness and/or the perceived spectral balance of an audio signal |
US8199933B2 (en) | 2004-10-26 | 2012-06-12 | Dolby Laboratories Licensing Corporation | Calculating and adjusting the perceived loudness and/or the perceived spectral balance of an audio signal |
WO2007120452A1 (fr) | 2006-04-04 | 2007-10-25 | Dolby Laboratories Licensing Corporation | Mesure et modification de la sonie d'un signal audio dans le domaine mdct |
TWI517562B (zh) | 2006-04-04 | 2016-01-11 | 杜比實驗室特許公司 | 用於將多聲道音訊信號之全面感知響度縮放一期望量的方法、裝置及電腦程式 |
ES2359799T3 (es) | 2006-04-27 | 2011-05-27 | Dolby Laboratories Licensing Corporation | Control de ganancia de audio usando detección de eventos auditivos basada en la sonoridad específica. |
CA2665153C (fr) | 2006-10-20 | 2015-05-19 | Dolby Laboratories Licensing Corporation | Traitement dynamique audio utilisant une reinitialisation |
US8521314B2 (en) | 2006-11-01 | 2013-08-27 | Dolby Laboratories Licensing Corporation | Hierarchical control path with constraints for audio dynamics processing |
FR2916079A1 (fr) * | 2007-05-10 | 2008-11-14 | France Telecom | Procede de codage et decodage audio, codeur audio, decodeur audio et programmes d'ordinateur associes |
ATE535906T1 (de) | 2007-07-13 | 2011-12-15 | Dolby Lab Licensing Corp | Tonverarbeitung mittels auditorischer szenenanalyse und spektraler asymmetrie |
WO2009086174A1 (fr) | 2007-12-21 | 2009-07-09 | Srs Labs, Inc. | Système pour ajuster la sonie perçue de signaux audio |
US8538042B2 (en) | 2009-08-11 | 2013-09-17 | Dts Llc | System for increasing perceived loudness of speakers |
US9312829B2 (en) | 2012-04-12 | 2016-04-12 | Dts Llc | System for adjusting loudness of audio signals in real time |
Citations (2)
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US5570363A (en) * | 1994-09-30 | 1996-10-29 | Intel Corporation | Transform based scalable audio compression algorithms and low cost audio multi-point conferencing systems |
US6006173A (en) * | 1991-04-06 | 1999-12-21 | Starguide Digital Networks, Inc. | Method of transmitting and storing digitized audio signals over interference affected channels |
-
2001
- 2001-02-02 FR FR0101662A patent/FR2820573B1/fr not_active Expired - Fee Related
-
2002
- 2002-01-31 WO PCT/FR2002/000366 patent/WO2002063609A1/fr active IP Right Grant
- 2002-01-31 EP EP02701358A patent/EP1356455B1/fr not_active Expired - Lifetime
- 2002-01-31 DE DE60224005T patent/DE60224005T2/de not_active Expired - Lifetime
Patent Citations (2)
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US6006173A (en) * | 1991-04-06 | 1999-12-21 | Starguide Digital Networks, Inc. | Method of transmitting and storing digitized audio signals over interference affected channels |
US5570363A (en) * | 1994-09-30 | 1996-10-29 | Intel Corporation | Transform based scalable audio compression algorithms and low cost audio multi-point conferencing systems |
Non-Patent Citations (2)
Title |
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BENJELLOUN TOUIMI A ET AL: "A SUMMATION ALGORITHM FOR MPEG-1 CODED AUDIO SIGNALS: A FIRST STEP TOWARDS AUDIO PROCESSED DOMAIN UN ALGORITHME DE SOMMATION DES SIGNAUX AUDIO CODES MPEG-1: PREMIEREETAPE VERS LE TRAITEMENT AUDIO DANS LE DOMAINE COMPRESSE", ANNALES DES TELECOMMUNICATIONS - ANNALS OF TELECOMMUNICATIONS, PRESSES POLYTECHNIQUES ET UNIVERSITAIRES ROMANDES, LAUSANNE, CH, vol. 55, no. 3/4, March 2000 (2000-03-01), pages 108 - 116, XP000948703, ISSN: 0003-4347 * |
FOO S ET AL: "ENHANCING THE QUALITY OF LOW BIT-RATE REAL-TIME INTERNET COMMUNICATIO SERVICES", INTERNET RESEARCH: ELECTRONIC NETWORKING APPLICATIONS AND POLICY, XX, XX, vol. 9, no. 3, 1999, pages 212 - 224, XP000901290, ISSN: 1066-2243 * |
Cited By (4)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US9202256B2 (en) | 2003-06-13 | 2015-12-01 | The Nielsen Company (Us), Llc | Methods and apparatus for embedding watermarks |
US9191581B2 (en) | 2004-07-02 | 2015-11-17 | The Nielsen Company (Us), Llc | Methods and apparatus for mixing compressed digital bit streams |
US8972033B2 (en) | 2006-10-11 | 2015-03-03 | The Nielsen Company (Us), Llc | Methods and apparatus for embedding codes in compressed audio data streams |
US9286903B2 (en) | 2006-10-11 | 2016-03-15 | The Nielsen Company (Us), Llc | Methods and apparatus for embedding codes in compressed audio data streams |
Also Published As
Publication number | Publication date |
---|---|
EP1356455B1 (fr) | 2007-12-12 |
FR2820573B1 (fr) | 2003-03-28 |
EP1356455A1 (fr) | 2003-10-29 |
DE60224005T2 (de) | 2009-01-22 |
FR2820573A1 (fr) | 2002-08-09 |
DE60224005D1 (de) | 2008-01-24 |
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