MX355274B

Audio decoder, apparatus for generating encoded audio output data and methods permitting initializing a decoder.

Abstract

An audio decoder decodes a bit stream of encoded audio data, wherein the bit stream of encoded audio data represents a sequence of audio sample values and comprises a plurality of frames, wherein each frame includes associated encoded audio sample values. The audio decoder comprises a determiner configured to determine whether a frame of the encoded audio data is a special frame comprising encoded audio sample values associated with the special frame and additional information, wherein the additional information comprise encoded audio sample values of a number of frames preceding the special frame, wherein the encoded audio sample values of the preceding frames are encoded using the same codec configuration as the special frame, wherein the number of preceding frames is sufficient to initialize the decoder to be in a position to decode the audio sample values associated with the special frame if the special frame is the first frame upon start-up of the decoder. The decoder comprises an initializer configured to initialize the decoder, wherein initializing the decoder comprises decoding the encoded audio sample values included in the additional information before decoding the encoded audio sample values associated with the special frame.

MX355274B, drawing sheet 1 — Sheet 1 of 76

Term

8.1 yearsleft in the term

Expires 14 October 2034.

20 claims: 6 independent, 14 dependent

1
REIVINDICACIONES 1. Un decodiflcador de audio para decodificar una corriente de bits de datos de audio codificado, caracterizado porque la corriente de bits de datos de audio codificado representa una secuencia de valores de muestras de audio y comprende una pluralidad de tramas, en donde cada trama incluye valores de muestras de audio codificado asociados, el decodificador de audio comprende:un determinador configurado para determinar si una trama de los datos de audio codificado es una trama especial que comprende valores de muestras de audio codificado asociados con la trama especial e información adicional, en donde la información adicional comprende valores de muestras de audio codificado de un número de tramas que preceden la trama especial, en donde los valores de muestras de audio codificado de las tramas precedentes se codifican usando la misma configuración de códec que la trama especial, en donde el número de tramas precedentes, que corresponde a tramas de premontaje, corresponde al número de tramas necesarias por el decodificador para acumular la señal completa durante el arranque del decodificador de manera que se encuentre en una posición para decodificar los valores de muestras de audio asociados con la trama especial si la trama especial es la primera trama tras el arranque del decodificador;y un inicializador configurado INSTITUTO DI LA ?aOF’KL AD para inicializar el decodificador si el. determinador determina que la trama es una trama especial, en donde inicializar el decodificador comprende decodificar los valores de muestras de audio codificado incluidos en la información adicional antes de la decodificación de los valores de muestras de audio codificado asociados con la trama especial, en donde el inicializador se configura para conmutar el decodificador de audio de una configuración de códec actual a una configuración de códec diferente, si el determinador determina que la trama es una trama especial y si los valores de muestra de audio de la trama especial se han codificado usando la configuración de códec diferente, y en donde el decodificador se configura para decodificar la trama especial usando la configuración de códec actual, y para descartar la información adicional si el determinador determina que la trama es una trama especial y si los valores de muestra de audio de la trama especial se han codificado usando la configuración de códec actual.
2
El decodificador de audio de conformidad con la reivindicación 1, caracterizado además porque la información adicional comprende información sobre la configuración de códec utilizada para codificar los valores de muestras de audio asociados con la trama especial, en donde el determinador se configura para determinar si la configuración de códec de la información adicional es . *L . J,\ST diferente de la configuración de códec actual.
3
El decodificador de audio de conformidad con una de las reivindicaciones 1 y 2, caracterizado además porque comprende un dispositivo de fundido cruzado configurado para llevar a cabo el fundido cruzado entre una pluralidad de valores de muestras de salida obtenidos usando la configuración de códec actual, y una pluralidad de valores de muestras de salida obtenidos al decodificar los valores de muestras de audio codificado asociados con la trama especial.
4
El decodificador de audio de conformidad con la reivindicación 3, caracterizado además porque el dispositivo de fundido cruzado se configura para llevar a cabo el fundido cruzado de valores de muestras de salida obtenidos al vaciar el decodificador en la configuración de códec actual y valores de muestras de salida obtenidos al decodificar los valores de muestras de audio codificado asociados con la trama especial.
5
El decodificador de audio de conformidad con una de las reivindicaciones 1 a 4, caracterizado además porque una trama más temprana del número de tramas comprendidas en la información adicional no se codifica de manera diferencial en tiempo, o se codifica por entropía, en relación con cualquier trama previa a la trama más temprana, y en donde la trama especial no se codifica de manera diferencial en tiempo o se codifica po'feritropiaen relación con cualquier trama previa a la 1 ’tTaludiftásTeirt|5f aba del número de tramas que preceden la trama especial o en relación con cualquier trama previa a la trama especial.
6
El decodificador de audio de conformidad con una de las reivindicaciones 1 a 5, caracterizado además porque la trama especial comprende la información adicional como una carga útil de extensión y en donde el determinador se configura para evaluar la carga útil de extensión de la trama especial.
7
Un aparato para generar una corriente de bits de datos de audio codificado que representa una secuencia de valores de muestras de audio de una señal de audio, caracterizado porque la corriente de bits de datos de audio codificado comprende una pluralidad de tramas, en donde cada trama incluye valores de muestras de audio codificado asociados, en donde el aparato comprende:un proveedor de trama especial configurado para proporcionar al menos una de las tramas como una trama especial, la trama especial comprende valores de muestras de audio codificado asociados con la trama especial e información adicional, en donde la información adicional comprende valores de muestras de audio codificado de un número de tramas que preceden la trama especial, en donde los valores de muestras de audio codificado de las tramas precedentes se codifican usando la misma configuración de códec que la trama especial, y en donde el número de tramas precedentes, que corresponde a las tramas de pre-montaje, corresponde al número de tramas necesarias por un decodificador para acumular la señal completa durante el arranque del decodificador de manera que se encuentre en una posición para decodificar los valores de muestras de audio asociados con la trama especial si la trama especial es la primera trama tras el arranque del decodificador;y una salida configurada para sacar la corriente de bits de datos de audio codificado, en donde los datos de audio codificados comprenden una pluralidad de segmentos, en donde cada segmento se asocia con una de una pluralidad de porciones de la secuencia de valores de muestras de audio y comprende una pluralidad de tramas, en donde el adicionador de trama especial se configura para añadir una trama especial al comienzo de cada segmento independientemente de si la configuración de códec cambia o no.
8
El aparato de conformidad con la reivindicación 7, caracterizado además porque la información adicional comprende información sobre la configuración de códec utilizada para codificar los valores de muestras de audio asociados con la trama especial.
9
El aparato de conformidad con una de las reivindicaciones 7 u 8, el aparato caracterizado además Τ ·, r'TJ-Yl Λ, 3 INvUSTKIAl porque comprende:un proveedor de segmentos configurado para proporcionar segmentos asociados con diferentes porciones de la secuencia de valores de muestras de audio y codificados por diferentes configuraciones de codees, en donde el proveedor de trama especial se configura para proporcionar una primera trama de al menos uno de los segmentos como la trama especial;y un generador configurado para generar los datos de salida de audio al disponer el al menos uno de los segmentos siguiendo otro de los segmentos.
10
El aparato de conformidad con la reivindicación 9, caracterizado además porque el proveedor de segmentos se configura para seleccionar una configuración de códec para cada segmento basado en una señal de control.
11
El aparato de conformidad con la reivindicación 9 ó 10, caracterizado además porque el proveedor de segmentos se configura para proporcionar m versiones codificadas de la secuencia de valores de muestras de audio, con m á 2, en donde las m versiones codificadas se codifican usando diferentes configuraciones de codees, en donde cada versión codificada comprende una pluralidad de segmentos que representa la pluralidad de porciones de la secuencia de valores de muestras de audio, en donde el proveedor de trama especial se configura para proporcionar una trama especial al comienzo de cada uno de los segmentos.
12
El aparato de conformidad con la reivindicación 11, caracterizado además porque el proveedor de segmentos comprende una pluralidad de codificadores, cada uno se configura para codificar al menos en parte la señal de audio de acuerdo con una de la pluralidad de diferentes configuraciones de codees.
13
El aparato de conformidad con la reivindicación 12, caracterizado además porque el proveedor de segmentos comprende una memoria que almacena las m versiones codificadas de la secuencia de valores de muestras de audio.
14
El aparato de conformidad con las reivindicaciones 9 a 13, caracterizado además porque el proveedor de trama especial se configura para proporcionar la información adicional como una carga útil de extensión de la trama especial.
15
Un método para decodificar una corriente de bits de datos de audio codificado, caracterizado porque la corriente de bits de datos de audio codificado representa una secuencia de valores de muestras de audio y comprende una pluralidad de tramas, en donde cada trama incluye valores de muestras de audio codificado asociados, que comprende:determinar si una trama de los datos de audio l codificado es una trama especial que comprende valores de muestras de audio codificado asociados con la trama especial e información adicional, en donde la información adicional comprende valores de muestras de audio codificado de un número de tramas que preceden la trama especial, en donde los valores de muestras de audio codificado de las tramas precedentes se codifican usando la misma configuración de códec que la trama especial, en donde el número de tramas precedentes, que corresponde a tramas de premontaje, corresponde al número de tramas necesarias por un decodificador para acumular la señal completa durante el arranque del decodificador de manera que se encuentre en una posición para decodificar los valores de muestras de audio asociados con la trama especial si la trama especial es la primera trama tras el arranque del decodificador;inicializar el decodificador si se determina que la trama es una trama especial, en donde la inicialización comprende decodificar los valores de muestras de audio codificado incluidos en la información adicional antes de decodificar los valores de muestras de audio codificado asociados con la trama especial;conmutar el decodificador de audio de una configuración de códec actual a una configuración de códec diferente, si se determina que la trama es una trama especial y si los valores de muestras de audio de la trama especial se han codificado usando la configuración de códec diferente;y decodificar la trama especial usando la configuración de códec actual y descartar la información adicional si se determina que la trama es una trama especial, y si los valores de muestras de audio de la trama especial se han codificado usando la configuración de códec actual.
16
El método de conformidad con la reivindicación 15, caracterizado además porque la corriente de bits de datos de audio comprende un primer número de tramas codificadas usando una primera configuración de códec, y un segundo número de tramas que siguen al primer número de tramas y codificadas usando una segunda configuración de códec, en donde la primera trama del segundo número de tramas es la trama especial.
17
El método de conformidad con una de las reivindicaciones 15 ó 16, caracterizado además porque la información adicional comprende información sobre la configuración de códec utilizada para codificar los valores de muestras de audio asociados con la trama especial, el método comprende determinar si la configuración de códec de la información adicional es diferente de la configuración de códec actual usando aquellos valores de muestras de audio codificado de tramas en la corriente de bits, los cuales preceden la trama especial, se codifican.
18
Un método para generar una corriente de bits de datos de audio codificado que represen^~ws-a__s£rkiuuci¿. de valores de muestras de audio de una señal de audio, caracterizado porque la corriente de bits de datos de audio codificado comprende una pluralidad de tramas, en donde cada trama incluye valores de muestras de audio codificado asociados, que comprende:proporcionar al menos una de las tramas como una trama especial, la trama especial comprende valores de muestras de audio codificado asociados con la trama especial e información adicional, en donde la información adicional comprende valores de muestras de audio codificado de un número de tramas que preceden la trama especial, en donde los valores de muestras de audio codificado de las tramas precedentes se codifican usando la misma configuración de códec que la trama especial, y en donde el número de tramas precedentes, que corresponde a tramas de premontaje, corresponde al número de tramas necesarias por el decodificador para acumular la señal completa durante el arranque del decodificador de manera que se encuentre en una posición para decodificar los valores de muestras de audio asociados con la trama especial si la trama especial es la primera trama tras el arranque del decodificador;y generar la corriente de bits por medio de la concatenación de la trama especial y las otras tramas de la pluralidad de tramas, en donde los datos de audio codificados comprenden una pluralidad de segmentos, en donde cada segmento se asocia con una de una pluralidad de porciones de la secuencia de valores de muestras de audio y comprende una pluralidad de tramas, en donde una trama especial se agrega al comienzo de cada 5 segmento independientemente de si la configuración de códec cambia o no.
19
El método de conformidad con la reivindicación 18, caracterizado además porque la información adicional comprende información sobre la 10 configuración de códec utilizado para codificar los valores de muestras de audio asociados con la trama especial.
20
Un medio de almacenamiento no transitorio, caracterizado porque comprende el método como el que se reclama en una de las reivindicaciones 15 a 19.

Independent claims20

424 paragraphs in 19 sections, as filed

(54) Title: AUDIO DECODER, DEVICE FOR GENERATING CODED AUDIO OUTPUT DATA, AND METHODS THAT ALLOW THE INITIALIZATION OF A DECODER.

(54) Title: AUDIO DECODER, APPARATUS FOR GENERATING ENCODED AUDIO OUTPUT DATA AND METHODS PERMITTING INITIALIZING A DECODER.

(57) Summary

An audio decoder decodes an encoded audio data bitstream, where the encoded audio data bitstream represents a sequence of audio sample values and comprises a plurality of frames, where each frame includes sample values associated encoded audio. The audio decoder comprises a determiner configured to determine whether a frame of the encoded audio data is a special frame comprising encoded audio sample values associated with the special frame, and additional information, wherein the additional information comprises sample values. audio encoded from a number of frames preceding the special frame, where the encoded audio sample values of the preceding frames are encoded using the same codec configuration as the special frame, where the number of preceding frames is sufficient to initialize the decoder to be in a position to decode the sample values of audio associated with the special frame if the special frame is the first frame, with decoder startup. The decoder comprises an initializer configured to initialize the decoder, wherein decoder initialization comprises decoding the encoded audio sample values included in the additional information before decoding the encoded audio sample values associated with the special frame.

(57) Abstract

An audio decoder decodes a bit stream of encoded audio data, where the bit stream of encoded audio data represents a sequence of audio sample values and comprises a plurality of trames, where each trame ineludes associated encoded audio sample values. The audio decoder comprises a determiner configured to determine whether a trame of the encoded audio data is a special trame comprising encoded audio sample values associated with the special trame and additional Information, where the additional Information comprise encoded audio sample values of a number of trames preceding the special trame, where the encoded audio sample valúes of the preceding trames are encoded using the same codee configuration as the special trame, Where the number of preceding trames is sufficient to initialize the decoder to be in a position to decode the audio sample valúes associated with the special trame if the special trame is the first trame upon start-up of the decoder. The decoder comprises an initializer configured to initialize the decoder, where initializing the decoder comprises decoding the encoded audio sample values included in the additional Information before decoding the encoded audio sample values associated with the special trame.

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PATENT TITLE No. 355274

Headlines):

Home:

Denomination:

ZUR FORDERUNG der angewandten

Hansastrasse 27C, 80686, München, GERMANY

FRAUNHOFER-GESELLSCHAFT FORSCHUNG EV

AUDIO DECODER, APPARATUS TO GENERATE CODED AUDIO OUTPUT DATA, AND ^ METHODS THAT ALLOW INITIALIZATION OF A DECODER ^Λ i ¡11 I k

Gil

Classification:

Inventor (s):

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Number:

MX / a / 20 ^ BQ48 | ^^ pw

Validity: years M

Date of Vóa ^ llhien '

Exi Date

The referei patent

In accordance with the article from the date of presenti

Who subscribes to the present title lot) (Official Gazette of the Federation 25/01/2006, 06/05 / 2009,06 / 01/2010, · __

Regulations of the Mexican Institute of 'articles 1, 3, 4, 5, section V, subsection a) J 12/27/1999, amended on 10/10/2002, 07/29/20

Deputy Generals, Coordinator, Directors Dlv ______ stnr-'tif- - -s ··

Departmental and other subordinates of the Mexican InstituteTe lá JffllffiriW.I'TlMirti 08/04/2004 and 09/13/2007). T r <NfiÓENDORF; TEF?

NIKOLAUS FKn; DÓHLA; NIKOLAUS iteritfteiónal:

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Industrial.

Non-extendable, counted to rights.

Industrial Property Law 999. 01/26/2004, 06/16/2005, a), 4th and 12th fractions I and III of 7/2004, 07/28/2004 and 09/07/2007);

lexicon of Industrial Property (DOF írdo that delegates powers to the Divisional Directors Subdirectors, Coordinators 5/12/1999, amended on 02/04/2000, 07/29/2004,

This letter is signed with an advanced electronic signature (FIEL), based on articles 7 BIS 2 of the Industrial Property Law; 3 of its Regulations, and 1 section III, 2 section V, 26 BIS and 26 TER of the Agreement establishing the guidelines for the use of the Electronic Payment and Services Portal (PASE) of the Mexican Institute of Industrial Property, in the procedures indicated.

THE DIVISIONAL DIRECTOR OF PATENTS

NAHANNY CANAL REYES

Original string:

NAHANNY MARISOL CANAL REYES | 00001000000403252793 | Tax Administration Service | 1695 || MX / 2018/30364 | MX / a / 2016/004845 | Patent title PCT | 1220 | RRGO | Page (s) | 1suD3mgLn / 1 Noli7GzjuRR24P + s =

Digital stamp:

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Arenal No. 550, Floor 1, Pueblo Santa María Tepepan, Xochimilco, 16020, Mexico City.

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MX / 2018/30364

355^

AUDIO DECODER, GEÑER APPARATUS & K'3SS1'UÜ CODED AUDIO OUTPUT, AND METHODS THAT ALLOW THE

INITIALIZATION OF A DECODER

FIELD OF THE INVENTION

The present invention relates to audio encoding / decoding and, in particular, to a data encoding and decoding procedure, which allows initialization of a decoder as may be required when switching between different code configurations [encoder-decoders ].

BRIEF DESCRIPTION OF THE INVENTION

The embodiments of the invention can be applied to scenarios in which the properties of the transmission channels can vary widely depending on the access technology, such as DSL [digital subscriber line], WiFi, 3G [ Third Generation], LTE [Long Term Evolution] and the like. Cell phone reception may fade indoors or in rural areas. The quality of wireless Internet connections depends heavily on the distance to the base station and access technology, leading to fluctuations in the bit rate. The available bit rate per user can also change with the

.. · '.'. ΊΟ 1 <number of clients connected to a tea'SO-station> -_

It is the object of the invention to provide a concept that enables the provision of audio content in a flexible manner.

In accordance with the invention, this object is achieved by an audio decoder according to claim 1, an apparatus for generating encoded audio output data according to claim 9, a method for decoding audio input data according to with claim 18, a method of generating encoded audio data according to claim

22, and a computer program according to claim 25.

The embodiments of the invention provide an audio decoder for decoding an encoded audio data bitstream, wherein the encoded audio data bitstream represents a sequence of audio sample values and comprises a plurality of frames, where each frame includes associated encoded audio sample values, the audio decoder comprises:

a determiner configured to determine whether a frame of the encoded audio data is a special frame comprising encoded audio sample values associated with the special frame and information

additional, where the additional information ^ comprises encoded audio sample values of \ a number of frames preceding the special frame, where the encoded audio sample values of the preceding frames are encoded using the same codec configuration as the frame special, where the number of preceding frames is sufficient to initialize the decoder to be in a position to decode the audio sample values associated with the special frame if the special frame is the first frame after decoder startup; and an initializer configured to initialize the decoder if the determiner determines that the frame is a special frame, wherein initializing the decoder comprises decoding the encoded audio sample values included in the additional information before decoding the associated encoded audio sample values. with the special plot.

The embodiments of the invention provide an apparatus for generating an encoded audio data bit stream representing a sequence of audio sample values of an audio signal, wherein the encoded audio data bit stream comprises a plurality of frames, where each frame includes associated encoded audio sample values, where

the apparatus comprises:

a special frame provider configured to provide at least one of the frames as a special frame, the special frame comprises encoded audio sample values associated with the special frame and additional information, wherein the additional information comprises audio sample values encoded from a number of frames preceding the special frame, where the encoded audio sample values of the preceding frames are encoded using the same codec configuration as the special frame, and where the number of preceding frames is sufficient to initialize a decoder to be in a position to decode the values of audio samples associated with the special frame if the special frame is the first frame after decoder startup; and an output configured to produce the encoded audio data bitstream.

The embodiments of the invention provide a method for decoding an encoded audio data bitstream, wherein the encoded audio data bitstream represents a sequence of audio sample values and comprises a plurality of frames, wherein each frame includes associated encoded audio sample values, comprising:

ii; íí, i'j-ic -Μ ·. '· .ΓΛΓ'Ο n í /. 27. JAL determine whether a frame of the encoded data is a special frame comprising encoded audio sample values associated with the special frame and additional information, wherein the additional information comprises encoded audio sample values of a number of frames. preceding the special frame, where the encoded audio sample values of the preceding frames are encoded using the same codec setting as the special frame, wherein the number of preceding frames is sufficient to initialize a decoder to be in a position to decode the audio sample values associated with the special frame if the special frame is the first frame after decoder startup; and initializing the decoder if the frame is determined to be a special frame, wherein initialization comprises decoding the encoded audio sample values included in the additional information before decoding the encoded audio sample values associated with the special frame.

The embodiments of the invention provide a method of generating an encoded audio data bit stream representing a sequence of audio sample values of an audio signal, wherein the encoded audio data bit stream comprises a plurality of frames, in audio sample values includes:

associated encoded, which provide at least one of the frames as a special frame, the special frame comprises encoded audio sample values associated with the special frame and additional information, wherein the additional information comprises encoded audio sample values of a number of frames preceding the special frame, where the encoded audio sample values of the preceding frames are encoded using the same codec configuration as the special frame, and where the number of preceding frames is sufficient to initialize a decoder to be in a position to decode the values of audio samples associated with the special frame if the special frame is the first frame after decoder startup; and generating the bit stream by concatenating the special frame and the other frames of the plurality of frames.

The embodiments of the invention are supported by the finding that the immediate response of an encoded audio data bitstream representing a sequence of audio sample values of an audio signal and comprising a plurality of frames may ι μ ρ τ r;

Λ -Λ. · '·' · \ '\ Achieved if one of the frames is provided as a special frame that includes encoded audio sample values associated with preceding frames, which are required to start a decoder to be in a position to decode the values of coded audio samples associated with the special frame. The number of frames required to start the decoder, therefore, depends on the codec configuration used, and is known for the code configurations. The modes of the invention are supported by the finding that switching between different code configurations can be beneficially achieved if said special frame is arranged in a position where switching between coding configurations will take place. The special frame may not only include encoded audio sample values associated with the special frame, but also information that enables switching between code configurations and immediate response with switching. In embodiments of the invention, the apparatus and method for generating encoded audio output data and the audio encoder are configured to prepare encoded audio data such that immediate response can occur with switching between code configurations, on the decoder side. In embodiments of the invention, said audio data

generated and output from the encoder side are received as audio input data from the decoder side, and allow immediate response from the decoder side. In embodiments of the invention, immediate response is allowed on the decoder side with switching between different code sets on the decoder side.

In embodiments of the invention, the initializer is configured to switch the audio decoder from a current codec configuration to a different codec configuration, if the determiner determines that the frame is a special frame and if the audio sample values of the Special frames have been encoded using different codec settings.

In embodiments of the invention, the decoder is configured to decode the special frame using the current codec configuration, and to discard the additional information if the determiner determines that the frame is a special frame and if the audio sample values of the frame special ones have been encoded using the current codec settings.

In embodiments of the invention, the additional information comprises information about the codec configuration used to encode the audio sample values associated with the special frame, where the

DL L /.

determiner is configured to determine if the codec setting for the additional information is different from the current codec setting.

In embodiments of the invention, the audio decoder comprises a crossfade device configured to perform crossfade between a plurality of output sample values obtained using the current codec configuration, and a plurality of output sample values obtained by decoding the encoded audio sample values associated with the special frame. In embodiments of the invention, the crossfade device is configured to crossfade the output sample values obtained by emptying the decoder in the current codec setting and the output sample values obtained by decoding the values. of coded audio samples associated with the special frame.

In embodiments of the invention, an earlier frame of the number of frames comprised in the additional information is not time-differentially encoded, or entropy-coded, relative to any frame prior to the earlier frame, and where the special frame is not time differential encoded or entropy encoded relative to any frame prior to the earliest frame of frame number

-C-

that precede the special plot or in relation to any plot prior to the special plot.

In embodiments of the invention, the special frame comprises the additional information as an extension payload and where the determiner is configured to evaluate the extension payload of the special frame. In embodiments of the invention, the additional information comprises information on the codec configuration used to encode the audio sample values associated with the special frame.

In embodiments of the invention, the encoded audio data comprises a plurality of segments, where each segment is associated with one of a plurality of portions of the audio sample value sequence and comprises a plurality of frames, where the adder Special Frame Set is configured to add a special frame at the beginning of each segment.

In one embodiment of the invention, the encoded audio data comprises a plurality of segments, wherein each segment is associated with one of a plurality of portions of the audio sample value sequence, and comprises a plurality of frames, in wherein the apparatus for generating an encoded audio data bitstream comprises a segment provider configured to provide segments associated with

different portions of the sequence of audio sample values and encoded by different code configurations, wherein the provider of the special frame is configured to provide a first frame of at least one of the segments as the special frame; and a generator configured to generate the audio output data by arranging at least one of the segments following another of the segments. In embodiments of the invention, the segment provider is configured to select a codec configuration for each segment based on a control signal. In embodiments of the invention, the segment provider is configured to provide m encoded versions of the audio sample value sequence, with m X 2, where the m encoded versions are encoded using different code configurations, where each version encoded comprises a plurality of segments representing the plurality of portions of the audio sample value sequence, wherein the provider of the special frame is configured to provide a special frame at the beginning of each of the segments.

In embodiments of the invention, the segment provider comprises a plurality of encoders, each configured to encode at least in part the audio signal in accordance with one of the plurality of

different code configurations. In embodiments of the invention, the segment provider comprises a memory that stores the encoded versions of the sequence of values of audio samples.

In embodiments of the invention, the additional information is presented in the form of a special frame extension payload.

In embodiments of the invention, the method for decoding comprises switching the audio decoder from a current codec configuration to a different codec configuration, if the frame is determined to be a special frame, and if the audio sample values of The special frame have been encoded using the different codec settings.

In embodiments of the invention, the encoded audio data bitstream comprises a first number of frames encoded using a first codec configuration, and a second number of frames following the first number of frames and encoded using a second codec configuration , where the first frame of the second number of frames is the special frame.

In embodiments of the invention, the additional information comprises information on the codec configuration used to encode the audio sample values associated with the special frame, and the method

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it comprises determining whether the codec configuration of the additional information is different from the current codec configuration using those bitstream stream encoded audio sample values, which precede the special frame, are encoded.

In embodiments of the invention, the method of generating an encoded audio data bitstream comprises providing segments associated with different portions of the audio sample value sequence and encoded by different code configurations, wherein a first frame of per at least one of the segments is provided as the special plot.

Thus, in embodiments of the invention, crossfade is performed to allow continuous switching between different code configurations. In embodiments of the invention, the additional information of the special frame comprises the pre-assembly frames necessary to initialize a decoder to be in a position to decode the special frame. In other words, in embodiments of the invention, the additional information comprises a copy of those encoded audio sample value frames preceding the special frame and encoded using the same codec configuration as the encoded audio sample values represented by the special plot

required to initialize the decoder to be in position to decode the audio sample values associated with the special frame.

In embodiments of the invention, special frames are input to encoded audio data with regular time intervals, ie, on a periodic basis. In embodiments of the invention, a first frame of each encoded audio data segment is a special frame. In modes, the audio decoder is configured to decode the special frames and subsequent frames using the codec setting indicated in the special frame until another special frame is found indicating a different codec setting.

In embodiments of the invention, the decoder and the method for decoding are configured to perform crossfade when switching from one codec configuration to another codec configuration, to allow continuous switching between multiple compressed audio representations.

In embodiments of the invention, the different code configurations are different code configurations according to the AAC standard (Coding

Advanced Audio), that is, different code settings of the YAC / family codes. The embodiments of the invention can be directed to switching between

MLUSTRiAL AAC family code code configurations and AMR (Adaptive Multiple Speed) family code code configurations.

In this way, the embodiments of the invention allow immediate response on the decoder side and switching between different code configurations, so that the way in which the audio content is delivered can be adapted to the environmental conditions, such as a transmission channel with

<td>variable bits. Of</td><td>this way,</td><td>the modalities</td><td>of the</td>

<td>invention allow</td><td>provide</td><td colspan="2">to the consumer the best</td>

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<td>determined.</td><td></td><td></td><td></td>

<td>BRIEF</td><td>DESCRIPTION OF</td><td>THE FIGURES</td><td></td>

Next, the embodiments of the invention are described with reference to the attached figures, in which:

Figure 1 shows a schematic view of an embodiment of an apparatus for generating encoded audio output data;

Figure 2 shows a schematic view to explain an embodiment of a special frame;

Figure 3 shows a schematic view of different representations of an audio signal;

Figure 4a and Figure 4b show views

schematics of apparatus for generating output data from rr -—-— —tti »m encoded audio;

Figure 5 shows a schematic view of an audio decoder;

Figure 6 shows a schematic block diagram for explaining an embodiment of an audio decoder and a method for decoding;

Figure 7 shows a schematic block diagram to explain a switching of an audio decoder between different code configurations;

Figure 8 shows a schematic diagram to explain the behavior of the AAC decoder (Advanced Audio Coding);

Figure 9 shows the switching from a first stream 1 to a second stream 2; and Figure 10 shows an exemplary syntax element that provides additional information.

DETAILED DESCRIPTION OF THE INVENTION

Generally, the embodiments of the invention aim at the provision of audio content, possibly combined with the supply of video, on a transmission channel with variable data rate. The goal may be to provide the consumer with the best possible audio quality, for a given network condition. The embodiments of the invention focus on the implementation of AAC family codes in a continuous adaptive environment.

In embodiments of the invention, as used herein, the audio sample values that are not encoded represent time domain audio sample values, such as PCM (encoded pulse modulated) samples. In embodiments of the invention, the term "coded audio sample value" refers to frequency domain sample values after encoding the time domain audio sample values. In embodiments of the invention, the coded audio samples or sample values are those obtained by converting the time domain samples into a spectral representation, such as by means of a MDCT (modified discrete cosine transform) , and encode the result, such as by quantization and Huffman encoding. Therefore, in embodiments of the invention, encoding means obtaining the frequency domain samples from the time domain samples, and decoding means obtaining the time domain samples from the frequency domain samples. Sample (sample) values obtained by decoding encoded audio data are sometimes referred to herein as

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Figure 1 shows one embodiment of an apparatus for generating encoded audio output data. Figure 1 shows a typical adaptive audio stream scenario, to which the embodiments of the invention can be applied. An audio input signal 10 is encoded by various audio encoders 12, 14, 16, and 18, that is, encoders 1 to m. Encoders 1 am can be configured to encode audio input signal 10 simultaneously. Typically, 1 am encoders can be configured such that a wide bit rate range can be achieved. The encoders generate different representations, ie encoded versions 22, 24, 26 and 28, of the audio input signal 10, ie representations 1 to m. Each representation includes a plurality of segments 1 to k, where the second segment of the first representation has been given the reference number 30, for exemplary purposes only. Each segment comprises a plurality of frames (access units) designated by the letters AU, and a respective index 1 to n, which indicates the position of the frame in the respective representation. Reference number 40 is provided for the eighth frame of the first representation, for exemplary purposes only.

Encoders 12, 14, 16 and 18 are configured λ <ΡΙ /?^ Γ ·

X Λ. \ yyyy •;'-z'zz2y ·' ', Kz-ry' 'ΑΧΧΧλ'ι to insert current access points (SAP) 42, with regular time intervals, which define the sizes of the segments. In this way, a segment, such as segment 30, consists of multiple frames, such as AU5, AU6, AU7 and AU8, where the first frame, AU5, represents a SAP 42. In Figure 1, SAPs are indicated by shading. Each lam representation represents a compressed audio representation (CAR) for the audio input signal 10, and consists of k of such segments. Switching between different CARs can take place at segment edges.

On the decoder side, a customer can request one of the representations that is best suited for a given situation, for example, for certain network conditions. If for some reason the conditions change, the client must be able to request a different CAR, the apparatus to generate the encoded output data must be able to switch between different CARs at each segment boundary, and the decoder must be able to switch to decode the different CAR in each segment limit. Therefore, the client may be in a position to tailor the media bit rate to the available channel bit rate, to maximize quality while minimizing insufficiency in · »·. - · r * · 'ί · £ y. · ?.

memory% yy ^ yy ,, - yy, '¡, 1 yyyintermedia (buffering). If the

HTTP (Hypertext Transfer Protocol) to download the segments, such a stream architecture can be referred to as an adaptive HTTP stream.

Current implementations include

Apple's HTTP Live Streaming (HLS), Microsoft's Seamless Streaming, and Adobe's Dynamic Streaming, which all follow the basic principle. Recently, MPEG launched an open standard: Dynamic Adaptive Continuous Transmission over HTTP (MPEG DASH), see Guidelines for Implementation: DASH-AVC / 264 Interoperability Points, http: // dashif.org/w/2013/08/DASHAVC-264~v2.OQ-hd-mca.pdf. HTTP typically uses TCP / IP (Transmission Control Protocol / Internet Protocol) as the underlying network protocol. The embodiments of the invention can be applied to all these current developments.

A switch between representations (encoded versions) will be as continuous as possible. In other words, there will be no click or audible glitch during switching. Without other measures provided by the modalities of the invention, this requirement

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It can only be achieved under certain restrictions and if special care is taken during the encoding process.

In Figure 1, the respective encoder from which a segment originates is indicated by a respective mark placed within a circle. Figure 1 further shows a decision engine 50, which decides which representation to download for each segment. A generator 52 generates encoded audio output data 54 of the selected segments which have reference numbers 44, 46 and 48 determined in Figure 1, by concatenating the selected segments. The encoded audio output data 54 may be supplied to a decoder 60 configured to decode the encoded audio output data into an audio output signal 62 comprising audio output samples.

In the embodiment shown in Figure 1, the segments, and therefore the frames, which originate from different encoders, are fed into the same decoder 60, eg AU4, from encoder 2, and AU5, from encoder 3, in the example of Figure 1. In case the same decoder example is used to decode such AUs, it is necessary that both encoders are compatible with each other. In particular, without any additional measures, this

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procedure cannot work if the two encoders are from a completely different code family, say AMR for encoder 2e, and G.711 for encoder 3. However, even when the same codec is used across all representations, Special care must be taken to restrict the encoding process. This is because modern audio encodings, such as Advanced Audio Coding (7 \ AC), are flexible algorithms that can operate in various settings using various tools and encoding modes. Examples of such coding tools in AAC are Spectral Band Replication (SBR) or Short Blocks (SB). Other important configuration parameters are the sampling frequency (fs(for example 48 kHz) or channel settings (mono, stereo, surround). To decode (AU) frames correctly, the decoder must know what tools are used and how they are configured (for example, fs or SBR crossover frequency). Therefore, in general, the required information is encoded in a short configuration string, and made available to the decoder before decoding. These configuration parameters can be referred to as codec configuration. In the case of AAC, this configuration is known as

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Audio Specific Configuration (ASC). ----------- To date, to achieve continuous switching, it was necessary to restrict the codec settings to be compatible through representations (encoded versions). For example, the sampling rate or coding tools typically must be identical across all representations. If incompatible code configurations are used between the representations, then the decoder must be reconfigured. This basically means that the old decoder has to be closed, and that the new decoder must be started with a new configuration.

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immediately, but requires multiple pre-mount AUs to accumulate full signal strength. This startup behavior is typical of codes that have a decoder state, that is, where the decoding of the current AU is not completely independent of the decoding of previous AUs.

As a result of this behavior, the codec settings were typically required to be constant across all representations, and the

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The only changing parameter was the bit rate. This is, for example, the case for the DASH-AVC / 264 profile as defined by the DASH Industry Forum.

This restriction does limit the flexibility of the codec, and therefore the efficiency of encoding across the full bit rate range. For example, SBR is a value encoding tool for very low bit rates, but it limits audio quality to high bit rates. Therefore, if the encoded configuration was required to be constant, i.e. with or without SBR, you had to compromise on either the high or low bit rate. Similarly, the coding efficiency could benefit from changing the sampling rate across the plots, but had to be kept constant due to the limitations mentioned above for continuous switching.

The embodiments of the present invention are directed to a novel method that enables continuous audio switching in an adaptive continuous environment, and in particular, allowing continuous audio switching for AAC family audio codes in a continuous adaptive environment. The inventive procedure is designed to address all of the disadvantages that result from limitations on codec settings, as described above. The general objective υ ~ * '. .: Λ ϊ Tfí 9 ϊ (¢ ^ ¾, k J> .V? - '- ϊ - <Ζ ^ Ζ-Λ ircnwrc <-'->

η: ia η ζ> '»η,. s ι \ l is to have more flexibility in configuration through representations (encoded versions), such as encoding tools or sample rate, while continuous switching is still allowed or ensured.

The embodiments of the invention are based on the finding that the restrictions explained above can be overcome, and greater flexibility can be achieved by adding a special frame that carries additional information in addition to encoded audio sample values associated with the special frame, among others. encoded audio data frames, such as compressed audio representation (CAR). A compressed audio representation can be considered a piece of audio material (music, voice, ...) after compression by a lossless or lossless audio encoder, for example an AAC family audio encoder (AAC , HE-AAC, MPEG-D USAC, ...) with a constant total bit rate. In particular, the additional information in the special frame is designed to allow instantaneous broadcasting on the decoder side, even in the case of a switch between different codex configurations. In this way, the special frame can be referred to as an instant broadcast frame (IPF). The IPF is configured to compensate for the delay of

decoder startup, and is used to transmit audio information about previous frames along with data from the current frame.

An example of such an instant broadcast frame

IPF 80 is shown in Figure 2. Figure 2 shows a number of frames (access units) 40, numbered n-4 to n + 3. Each frame includes associated encoded audio sample values, i.e. encoded audio sample values of a specific number of time domain audio sample values from a sequence of time domain audio sample values representing a audio signal, such as audio input signal 10. For example, each frame may comprise encoded audio sample values representing 1024 time domain audio sample values, that is, the audio sample values of an uncoded audio signal. In Figure 2, frame n arranged between the preceding frame n-1 and the next frame n + 1 represents the special frame or IPF 80. The special frame 80 includes additional information 82. The additional information 82 includes the information 84 on the codec configuration, that is, information on the codec configuration used in encoding the data stream that includes the frames n-4 to n + 3, and therefore, information on the codec settings used to encode

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In the mode shown in Figure 2, a delay introduced by an audio decoder is assumed to have three frames; that is, it is assumed that three so-called pre-mount frames are necessary to construct the complete signal during startup of the audio decoder. Therefore, assuming that the current setting (codec setting) is known to the decoder, the decoder will normally have to initiate decoding in frame n-3 to produce valid samples in frame n. Therefore, in order to make the necessary information available to the decoder, the additional information comprises a number of encoded audio sample value frames preceding the special frame 80 and encoded using the codec setting 84 indicated in the information. 82 additional. This number of frames is indicated by reference number 86 in Figure 2. This number of frames 86 is required to initialize the decoder to be in a position to decode the audio sample values associated with special frame n. Accordingly, the information in frame 86 is duplicated and carried as part of special frame 80. In this way, this information

it is available to the decoder immediately with the switch to the data stream shown in Figure 2 in frame n. Without this additional information in frame n, neither codec configuration 84 nor frames n-3 to n-1 can be made available to the decoder after a switch. Adding this information to the special frame 80 allows the decoder to be initialized immediately, and therefore, the immediate emission with the switch to a data stream comprising the special frame. The decoder is configured such that such initialization and decoding of frame n can be performed within the available time window until the output samples obtained by decoding frame n have to occur.

During normal decoding, that is, without switching to a different codec configuration, only frame n is decoded, and the frames included in the additional information, n-3 to n-1, are ignored. However, after switching to a different codec configuration, all the information in the special frame is extracted, and the decoder is initialized based on the included codec configuration, and based on the decoding of the three frames pre-montage (n-3 to n-1) before final decoding and replay of current frame n. Decoding the

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Pre-mount frames take place before the current frame is decoded and played again. The pre-mount frames are not played again, although the decoder is configured to decode the pre-mount frames within the available time window before replay of the current frame n.

The term codec configuration refers to the codec configuration used in encoding audio data or audio data frames. In this way, the encoding configuration may indicate different encoding tools and modes used, where the exemplary encoding tools used in AAC are spectral band replication (SBR) or short blocks (SB). A configuration parameter can be the crossover frequency SBR. Other configuration parameters may be the sample rate or the channel configuration. Different code configurations differ in one or more of these configuration parameters. In embodiments of the invention, different code configurations may also comprise completely different code such as AAC, AMR, or G.711.

Therefore, in the example illustrated in Figure 2, three frames, ie n-3 to n-1, are necessary to compensate for the decoder start delay.

Additional frame data can be transmitted over

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Τ 1 '· - •• «.τ *'» * · means of an extension payload mechanism within the audio bitstream For example, the extension payload USAC mechanism (UsacExtElement) can be used to port additional information. Additionally, the config field can be used to transmit the current 94 configuration. This may be useful in the case of bitstream switching or bit rate adaptation. Both the first pre-mount AU (n-3) and the IPF itself (η), can be an independently decodable frame. In the USAC context, encoders can set a flag (usacIndependencyFlag) to 1 for such frames. By implementing the frame structure shown in Figure 2, random access to the bitstream at each IPF and issuance of valid PCM samples are possible immediately. The decoding process of an IPF can include the following steps. All pre-mount AUs (n-3 ... n-1) are decoded and the resulting output PCM samples are discarded. The internal buffers and decoder states are fully initialized after this stage. Frame n is decoded, and regular broadcast starts. Decoding is continued in the normal way with the n + 1 frame. The IPF can be used as an Audio Stream Access Point (SAP). Immediate issuance possible

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of valid PCM samples at each IPF.

Special frames as defined herein can be implemented in any codec that allows multiplexing and transmission of auxiliary data or extension data, or data stream elements or similar mechanisms to transmit external data from the audio codec. The embodiments of the invention refer to the implementation for a USAC codec fabric. The embodiments of the invention can be implemented in connection with USAC audio encoders and decoders. USAC stands for Unified Audio and Voice Coding, and reference is made to the ISO / IEC 23003-3: 2012 standard. In embodiments of the invention, the additional information is contained in an extension payload of the corresponding frame, such as frame n in Figure 2. For example, the USAC standard allows the addition of arbitrary extension payload to data encoded audio. The existence of extension payload can be switched on a frame-by-frame basis. Consequently, the additional information can be implemented as a new type of extension payload defined to carry additional audio information from previous frames.

As explained above, instant broadcast frame 80 is designed in such a way that

Valid output samples associated with a certain timestamp (frame n) can be generated immediately, that is, without having to wait for the specific number of frames according to the delay of the audio codec. In other words, the delay of the audio codec can be compensated. In the mode shown in Figure 2, the audio codec delay is three frames. Furthermore, the IPF is designed in such a way that it can be fully and independently decoded, that is, without any other knowledge of the previous audio frame. In this sense, the earliest of the number of frames added to the special frame (i.e. the n-3 frame in Figure 2) is not time differential encoded or entropy encoded, relative to any previous frame. Furthermore, the special frame is not time differential encoded or entropy encoded relative to any frame prior to the earliest of the number of frames contained in the additional information or any previous frame. In other words, for n-3 and n frames in Figure 2, all dependencies to previous frames can be removed, for example, time differential encoding of certain parameters, or reentry of entropy encoding. Therefore, these independent frames allow the correct decoding and syntactic analysis of all the symbols, although, in themselves, they are not

enough to get valid PCM samples instantly. Although such independent frames are already available in common audio codes, such as AAC or USAC, such audio codecs do not provide special frames, such as IPF 80.

In embodiments of the invention, a special frame is provided at each current access point of the representations shown in Figure 1. In Figure 1, current access points are the first frame in each segment, and are shaded. Accordingly, Figure 1 shows a specific embodiment of an apparatus for generating encoded audio output data in accordance with the present invention. Still further, each of the 1 am encoders shown in Figure 1 represents one embodiment of an audio encoder in accordance with the invention. According to Figure

1, encoders 12-18 represent providers configured to provide segments associated with different portions of the audio input signal 10 and encoded by different code configurations. In this sense, each of the encoders 12 to 18 uses a different codec configuration. The decision unit 50 is configured to decide for each segment, the representation for the download. Therefore, the decision unit 50 is configured to select a configuration

codec (associated with the respective representation) for each segment based on a control signal. For example, the control signal may be received from a client that requires the representation that is best suited for a given situation.

On the basis of the decision of the decision unit 50, the block 52 generates the audio output data 54 by arranging segments one after the other, such as segment 46 (segment 2 of representation 3) following the segment 44 (segment 1 of representation 2). In this way, the special frame AU5 at the beginning of segment 2 allows switching to representation 3 and immediate edge response between segments 44 and 46 on the decoder side.

Thus, in the embodiment shown in Figure 1, a provider (comprising encoders 1 am) is configured to provide m encoded versions of audio input 10, where mk 2, where m encoded versions (representations ) are encoded using different code configurations, where each encoded version includes a plurality of segments representing the plurality of portions of the audio sample value sequence, where each of the segments comprises a special frame at its beginning.

In other embodiments of the invention, they can

Different representations of the same audio input, such as representations 22 to 28 in Figure 1, are stored in memory, and can be accessed if a user requires corresponding media content.

The 1 am encoder instances shown in Figure 1 may produce a different encoder delay depending on the encoder configuration and / or tool activation on the encoder instances. In such a case, measures can be taken to ensure that the encoder delays are compensated, to achieve a time alignment of the m output currents, ie the m representations. This can be implemented, for example, by adding a number of zero trace samples to the encoder input, to compensate for different encoder delays. In other words, the segments in the different representations will have the same duration to allow continuous switching between representations in the segment constraints. Theoretical segment durations depend on the sampling rates used, and the frame sizes. Figure 3 shows an example of possible insertion of IPF in representations with different framing, it may be due to different sampling rates and / or frame sizes. Zero samples can be added to shorter segments at an appropriate position,

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in such a way that all the special frames le-3 - s - e - enouer ^ -gej4 · aligned in time, as can be seen in Figure 3.

Figure 4a shows a schematic view of an apparatus 90 for generating encoded audio output data 102. Apparatus 90 comprises a provider 92 configured to provide at least one frame 80 of a plurality of frames 40, as defined by a special frame in the present application. In embodiments of the invention, provider 92 may be implemented as part of an encoder to encode the audio sample values, which provides frames 40 and adds the additional information to at least one of the frames, to generate the special frame. For example, provider 92 may be configured to add the additional information as a payload extension to one of frames 40 to generate special frame 80. Frames 40, 80, representing the encoded audio data bit stream 102 are produced by means of output 112.

Figure 4b shows a schematic view of an apparatus 100 for generating encoded audio output data 102. The apparatus comprises a provider 104 configured to provide segments 106, 108 associated with different portions of an audio sample value sequence. A first frame of at least one of the segments is a special frame, as explained with

anteriority. A generator 110 is configured to generate the audio output data by arranging at least one of the segments 106, 108, then another of the segments 106, 108. The generator 110 supplies the audio output data to the output 112 configured to output the encoded audio data 102.

Figure 5 shows a schematic view of one embodiment of the audio decoder 60 for decoding audio input data 122. The audio input data may be the output of block 52 shown in Figure 1. Audio decoder 60 comprises a determiner 130, an initializer 132 and a decoder core 134. Determiner 130 is configured to determine whether an audio input data frame 122 is a special frame. Initializer 132 is configured to initialize decoder core 134 if the frame is a special frame, and initialization is necessary or desired. Initialization comprises decoding the preceding frames included in the additional information. Decoder core 134 is configured to decode frames of encoded audio sample values using the codec setting with which it is initialized.

In the event that the frame is not a special frame, the decoder core 134 is supplied directly, arrow 136. In the event that the frame is

L IF A ') a special frame, and decoder core 134 initialization not required, determiner 130 may discard the additional information, and only supply the encoded audio sample values of the special frame (without the frames in the information additional) to the decoder core 134. Determiner 130 can be configured to determine whether initialization of decoder core 134 is necessary, based on the information included in the additional information, or based on the external information. The information included in the additional information may be information about the codec configuration used to encode the special frame, where the determiner may decide that initialization is necessary if the information indicates that the preceding frames are encoded using a different codec configuration such as the special plot. External information may indicate that decoder core 134 is to be initialized or reinitialized upon receipt of the next special frame.

In embodiments of the invention, the decoder is configured to start the decoder core 134 in one of different code configurations. For example, different instances of a software decoder core can be started using different codex settings, i.e. different codec settings parameter as explained above. In embodiments of the invention, decoder (core) initialization may comprise closing a current decoder instance and opening a new decoder instance using codec configuration parameters included in the additional information (i.e. within the bit stream received) or supplied externally, i.e. external to the received bit stream,

The decoder configurations of different ones can be switched to codees depending on the code configurations used to encode respective segments of the received encoded audio data.

Decoder 60 can be configured to switch from a current codec configuration, i.e., the codec configuration of the audio decoder before encountering the special frame, to a different codec configuration, if the additional information indicates a different codec configuration of the current codec settings.

Other details of an embodiment of an audio decoder having a decoder AAC behavior are explained with reference to Figures 6 to 8. Figure 8 schematically shows the

ί. ')>. ··· behavior of a 7 \ AC decoder. Reference is made to ISO / IEC DTR 14496-24, Audio Interaction and

Systems.

Figure 8 shows the behavior of the decoder over a number of states, a first state 200 corresponding to one or more pre-mount frames, a state associated with each of the AU1, AU2 and AU3 frames, and a clearing state 202.

To generate valid output samples for AUl, both the one or more pre-mount frames and the AUl frame must be decoded. The samples generated by the pre-mount frames are discarded, that is, they are used to initialize the decoder only, and are not played again. However, decoding of pre-mount frames is mandatory to establish internal decoder states. In embodiments of the invention, additional information on special frames includes pre-assembly frames. In this way, the decoder is in a position to decode the pre-mount frames to establish the internal states of the decoder so that the special frame can be decoded, and immediate issuance of valid output samples of the frame can take place special. The actual number of pre-mount AUs' (frames) depends on the decoder start delay, in the example in Figure 8,

an AU.

In general, for file playback,

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enough information to completely initialize the internal states of the decoder and fill the internal buffers. In this way, the insertion of special frames allows the immediate emission in positions of random currents.

The clearing state 202 in Figure 8 shows the behavior of the decoder if the clearing is performed after decoding the last AU3 frame. Emptying means feeding the decoder with a hypothetical zero frame, that is, a hypothetical frame made up of entirely digital zero input samples. Due to the addition of the AAC family overlay, flushing produces a valid output that is achieved without consuming a new input frame. This is possible because the last AU3 frame includes prediction information about the output sample values that will be obtained when decoding a next frame after the AU3 frame, since the frames overlap on a number of

time domain sample values. Generally, the first half of a frame overlaps with a preceding frame, and a second half of a frame overlaps with a subsequent frame. Therefore, the second half of the output sample values obtained when decoding a first frame includes information about the first half of the output sample values obtained when decoding a second frame after first plot. This feature can be exploited when a crossfade is implemented, as will be explained below.

Other details of an embodiment of an audio decoder and a method of decoding audio input data are now described with reference to the

Figure 6, where the audio decoder is configured to carry out the method described with reference to Figures 6 and 7. The process starts at 300. The decoder scans the incoming frames (AU) for an IPF, and determines if an incoming frame is an IPF, 302. If the incoming frame is not an IPF, the frame is decoded, 304, and the process jumps to the next frame, 306. If there is no next frame, the process ends. Decoded PCM samples are produced, as indicated by block 308, which may represent an output buffer. Yes ·· *! * '<&

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D ¿L? I 1 · '; Ο' Γ · Ο. · .Ο idcu is determined in 302 that the frame is an IP ^ r ~ ^ »« e5taJ-úa - codec setting, 310. For example, evaluates the config field shown in Figure 2. A determination is made in terms of whether the codec setting (stream setting) has changed, 312. If the codec setting did not change, i.e. if the additional information indicates a codec setting identical to the current codec setting, the additional information, such as the extension payload, is skipped, and the process jumps to 304, where the decoding is continued in the normal way.

If the codec settings have changed, the following steps apply. The decoder empties,

314. The output samples resulting from decoder emptying are stored in an emptying buffer, 316. These output samples (or at least a portion of these output samples) are a first input to a crossfade process 318. The decoder is then reinitialized using the new codec configuration, as indicated by the additional information, such as by the config field in Figure 2, and using the preceding frames comprised in the special frame. With the reset, the decoder is able to decode the special frame, i.e. the encoded audio sample values

VyhÍFyA ¡NL-USMUAL associated with the special plot. The toaran .a ^ ppHL is decoded, 322. The output samples (PCM samples) obtained by decoding the special frame are stored as a second input to the crossfade process 318. For example, the corresponding output samples PCMs can be stored in a buffer, 324, which can be called the IPF buffer. In crossfade process 318, a crossfade is calculated based on the two input signals, from the flush buffer and the IPF buffer. The crossfade result is produced as PCM output samples, block 308. The process then jumps to the next frame 306, and the process repeats for the next frame. In the event that the present frame is the last frame, the process ends.

Other details of such steps performed after a configuration change as detected in 312 are now explained with reference to Figure 7. The codec configuration is retrieved from the additional information of the IPF, 330, and provided for reset. of decoder 332. Before resetting the decoder, the decoder is emptied, 314, and the resulting output samples are stored in the dump buffer, 316. The ί

IN;

resetting the decoder can iricTuTr ^ eT ^ crSTTé * of the current decoder instance, and opening the new decoder instance, with the new settings. In reopening the new instance of the decoder, the information about the codec configuration contained in the IPF is used. After the new instance of the decoder is opened, it is initialized by decoding the pre-mount frames included in the IPF. The number of pre-mount frames contained in the IPF is assumed to be m, as indicated in block 334.

It is determined if m> 0, 336. If m> 0, the preassembly frame nm is decoded, 338, where n indicates the IPF. The output PCM samples obtained are discarded, 340. It is reduced m by one, and the process jumps to block 336. By repeating steps 336 to 342 for all pre-assembly frames contained in the IPF, a fill-in process is performed. the decoder states after the same reopening, 344. If all of the pre-mount frames have been decoded, the process jumps to block 332, where the IPF is decoded. The resulting PCM samples are supplied to the PCM buffer 342. Crossfade 318 is performed based on the outputs of the PCM buffers 316 and 324, and the output of the crossfade process 318 is fed into memory. intermediate output PCM 308.

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In the mode described above, resetting the decoder includes closing the current decoder instance, and opening a new decoder instance. In alternative embodiments, the decoder may include a plurality of decoder instances in parallel, so resetting the decoder may include switching between different instances of the decoder.

In addition, decoder reset includes filling the decoder states by decoding pre-mount frames included in the additional information of the special frame.

As explained above, by taking advantage of internal memory states and buffers (add overlay, filter states) in an AAC decoder, it is possible to obtain output samples without passing new input through the emptying process. The emptying output signal closely resembles the original signal for at least a part of the obtained output sample values, in particular its first part, see state 202 in Figure 8. The obtained output sample values by means of the casting process they are used for the cross melt process described in detail below.

As can be seen in state 202 in the

Figure 8, the energy in memory integme ^ ia— .. de. resulting voiding will decrease over time, according to the transform window and enabled tools of the current codec settings. Therefore, crossfade should be applied in the first part of the flush buffer, where the output signal can be considered almost full power. Exploiting the fact that modern audio codexes can be flushed out to obtain valid samples for a successive crossfade helps significantly in obtaining continuous switching values. Therefore, in embodiments of the invention, the crossfade device is configured to perform crossfade between output values obtained by a process of clearing the current codec setting, and the output sample values. obtained by decoding the special frame using the codec settings indicated in the additional information.

In the following, a specific embodiment of the crossfade process is described. Cross fade is applied to audio signals as described above, to avoid audible failure during CAR switching. A typical failure is a drop in the power of the output signal. As explained

INSTITUTE του tu;: · above, the energy of the emptied signal. ^ Ismijajjliá-.ü & ,. according to configuration. Thus, the intensity of the crossfade must be selected carefully, depending on the configuration, to avoid failure. If the crossfade window is too short, then the switching process may introduce audible faults due to the difference in the audio waveform. If the crossfade window is too long, then the emptied audio samples have already lost power and will cause a drop in the output signal power. For an AAC codec setup using short transformation windows of 256 samples, a linear crossfade with a length of n = 128 samples (per channel) can be applied. In other embodiments, a linear crossfade with a length of, for example, 64 samples (per channel) may be applied.

An example of a linear crossfade process using 128 samples is described below:

The crossfade process can use the first 128 samples from the flush buffer. The emptying buffer is displayed in a window by multiplying the first 128 samples of the emptying buffer Sf = Sfo, Sfi27 by

where i is the Index of the current sample. The result can be stored in an internal buffer of the crossfade device, i.e.sf = M * -¿) .....

Also, the IPF S bufferd It is shown in a window, where the first 128 decoded IPF output samples are multiplied by the factor i + l 128 'where i is the index of the current sample. The result can be stored in an internal buffer of the crossfade device, i.e.

, ..., S ^ 27 '1 / ···) $ dn ·

The first 128 samples from the internal buffers are added:

* 0 ~ Sd> 0 3 $ frO '·> $ (].' 127 + ·· Sdmr and the resulting values are output to buffer 308 of PCM output samples.

Therefore, linear crossfade is achieved on the first 128 output sample values from the flush buffer and the first 128 sample values from the IPF buffer.

Generally, the crossfade device can be configured to perform crossfade between a plurality of output sample values obtained using the current codec setting and a

Τ '3. you' and ~ 7 i í • 'vU

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^ ita £ 2 * SS 'plurality of output sample values obtained by decoding the encoded audio sample values associated with the special frame. In general, in audio codes, such as AAC family codes and AMR family codes, the encoded audio sample values of a preceding frame implicitly comprise information about the encoded audio signals in a subsequent frame. This property can be used in the implementation of crossfade when switching between different code configurations. For example, if the current codec setting is an AMR codec setting, the output sample values used in the crossfade may be obtained based on a zero impulse response, i.e. based on the response obtained when a zero frame is applied to the decoder core after the last frame of the current codec setting. In embodiments of the invention, additional mechanisms used in audio encoding and decoding, in crossfade, can be used. For example, the internal filters used in SBR (Spectral Band Replication) comprise delays, and therefore long settling times that can be used in crossfade. Therefore, the embodiments of the invention are not restricted to any ητ 'Si / τ

INSTITUTO MTXSCANO c: THE PROPERTY

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Specific crossfade to achieve continuous ”switching between code settings. For example, the crossfade device can be configured to apply increasing weights to a first number of the output sample values of the special screen, and to apply decreasing weights to a number of the output sample values obtained based on the decoding using the current codec setting, where weights can increase and decrease linearly, or can increase and decrease nonlinearly.

In embodiments of the invention, decoder initialization comprises initializing internal decoder states and buffers using the additional information in the special frames. In embodiments of the invention, decoder initialization takes place if the codec configuration changes. In other embodiments of the invention, the special frame can be used for decoder initialization without changing the codec configuration. For example, in embodiments of the invention, the decoder can be configured for immediate broadcast, where the internal states and buffers of a decoder are populated without changing a codec configuration, where crossfade can be performed with zero samples. In this way, the immediate issuance of

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Or i. Ι.Λ valid samples is possible. In other embodiments, a forward function can be implemented, where the special frame can be decoded at predetermined intervals according to the desired forward rate. In embodiments of the invention, the decision as to whether initialization using the special frame should take place, ie whether it is necessary or desired, can be made on the basis of an external control signal supplied to the audio decoder.

As explained above, the special frame (such as IPF 80 as shown in Figure 2) can be used for bit rate adaptation and bit rate switching, respectively. The following restrictions may apply: all representations (eg different bit rate, different use of encoding tools) are time aligned), IPFs are inserted into each representation; the IPFs are synchronized, and the config field of the IPF in Figure 2 contains the current settings, i.e. tool activation etc. Figure 9 shows an example of bit rate adoption by switching the bit stream in an adaptive stream environment. Control logic (such as the system shown in Figure 1), which is sometimes called a fabric, breaks the audio data into

I

segments. A segment includes mertipipie »'——, - Ja. Audio stream settings can change at each segment limit. The audio decoder is not aware of segmentation, it is only provided with simple AUs by control logic. To allow switching of the audio bitstream at each segment boundary, the first AU in each segment may be an IPF as explained above. In Figure 9, a segment limit 400 is indicated by the dotted line. In the scenario illustrated in Figure 9, the audio decoder is provided with AU 40 (AU1 to AU3) of Stream 1. The control logic decides to switch to Stream 2 at the next segment limit, i.e. the edge 400. After decoding AU3 from Stream 1, the control logic can pass AU4 from Stream 2 to the audio decoder, without any further warning. AU4 is a special frame (IPF), and therefore immediate broadcast can take place after switching to stream 2.

With reference to the scenario shown in Figure 9, the switching can take place as follows

<td>way:</td><td>for AU1 a</td><td>AU3 current</td><td>1, no</td><td>I know</td><td colspan="2">detects</td>

<td>IPF, and</td><td>the process</td><td>decoding it</td><td>carries</td><td>to</td><td>cape</td><td>of</td>

<td>way</td><td>normal. A</td><td>IPF is detected</td><td>for</td><td>AU4</td><td>of</td><td>the</td>

2. In addition, a change in the

audio

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INDVS 'current settings. The decoder initializes the emptying process, 4027 Figuraη Figure 9.

The resulting PCM output samples are stored in a temporary buffer (flush buffer) for later use. The audio decoder is reset with the current settings carried by the IPF. The IPF payload (preassembly) is decoded. The resulting PCM output samples are discarded. At this point, the internal decoder states and buffers are fully initialized. AU4 is decoded. To avoid fault switching, cross fade is applied. The PCM samples stored in the flush buffer are faded out, while the PCM samples resulting from the decoding of AU4 and stored in the PCM output buffer fade inbound. The crossfade result is output.

Therefore, the IPF can be used to allow the switching of compressed audio representations. The decoder can receive the simple AUs as input, and thus no other control logic is required.

Details of a specific modality are now described in the context of MPEG-D USAC, where the bitstream syntax can be as follows:

The syntax element— (4) is used to transmit audio information from previous frames along with data from the present frame. Additional audio data can be used to compensate for decoder start delay (pre-mount), thus allowing access Random on stream access points using AudioPreRoll (). A UsacExtElement () can be used to transmit the AudioPreRoll (). For this purpose, a new payload identifier will be used.

Table 1: Payload identifier for AudioPreRoll ().

<td>Name</td><td>Value</td>

<td>ID EXT ELE AUDIOPREROLL</td><td> 4</td>

The AudioPreRoll () syntax is shown in Figure 10 and explained in the following:

<td>configLen</td><td>element of</td><td>size syntax of</td><td>the</td>

<td></td><td>setting</td><td>in bytes.</td><td></td>

<td>Config ()</td><td>the element</td><td>syntax</td><td>the</td>

<td></td><td>setting</td><td>of the decoder. In</td><td>the</td>

<td></td><td>context of</td><td>MPEG-DUSAC this is</td><td>the</td>

<td></td><td>UseConfig ()</td><td colspan="2">as defined in ISO / IEC</td>

<td></td><td> 23003-3:2012.</td><td>The Config field</td><td> 0</td>

can be transmitted to be able to

Respond to changes in audio settings (current switching).

numPreRollFrames The number of pre-mount access units (AUs) transmitted as audio pre-mount data. The reasonable number of AUs depends on the decoder start delay.

auLen AU length in bytes.

Drive Access () the pre-assembly AU (s).

The pre-assembly data carried on the extension element may be transmitted out-of-band, that is, the requirements of the buffer may not be satisfied.

To use AudioPreRoll () for both random access and bitrate adaptation, the following restrictions apply:

- The first element of each frame is an extension element (UsacExtElement) of type id EXT ELE

AUDIOPREROLL.

The corresponding UsacExtElement () will be set as described in Table 2.

- Therefore, if pre-assembly data is presented, this UsacFrameO will start with the following bit sequence:

1: usacIndependencyFlag.

1: UsacExtElementPresent (to indicate pre-mount audio extension element).

0: UsacExtElementUseDefaultLength (to indicate audio pre-mount extension element).

- If no pre-mount data is transmitted, the extension payload will not be presented (UsacExtElementPresent = O).

The pre-mount frames with index 0 and numPreRollFrames-1 will be independently decodable, i.e. usacIndependencyFlag will be set to 1.

Table 2: Setting the UsacExtElement () to

AudioPreRoll ().

<td>UsacExtElementType</td><td>ID EXT ELE AUDIOPREROLL</td>

<td>UsacExtElementConfigLenght</td><td> 0</td>

<td>UsacExtElementDefaultLenghtPresent</td><td> 0</td>

<td>UsacExtElementPayloadFrag</td><td> 0</td>

Random access and immediate broadcast is possible in each frame using the AudioPreRoll () structure as described. The following pseudo-code describes the decoding process:

if (usacIndependencyFlag = 1) {

if (UsacExtElementPresent = 1 (/ * In this case UsacExtElementUseDefaultLength must be

0! * / if (UsacExtElementUseDefaultLength! = 0) go to error;

/ * Check for the presence of config and reinitialize if necessary * / int ConfigLen = getConfigLen (); if (ConfigLen> 0) {config c = getConfig (ConfigLen);

ReConfigureDecoder (c);

} / * Get the pre-mount AUs and decode, discard output samples * / int numPreRollFrames = getNumPreRollFrames (); for (auldx = 0; auldx <numPreRollFrames; auldx ++) int auLen = getAuLen ();

AU nextAU = getpreRollAU (auLen);

DecodeAU (nextAU);

}

/ * The decoder states are initialized at this point. Decoding * / is normally continued.

Bit rate adaptation can be used by switching between different encoded representations of the same audio content. The structure of

AudíoPreRoll () as described can be used for that purpose. The decoding process in case of bit rate adaptation is described by the following pseudocode:

if (usacIndependencyFlag = 1) {if (UsacExtElementPresent = 1 {/ * In this case UsacExtElementUseDefaultLength must be

0! * / if (UsacExtElementUseDefaultLength! = 0) go to error;

int ConfigLen = getConfigLen (); if (ConfigLen> 0) {config newConfig = getConfig (ConfigLen);

/ * Configuration did not change, skip AudioPreRoll and continue normal decoding * / if (newConfig == currentConfig) (

i ..

SkipAudioPreRoll ();

go to completion;

} / * Configuration changed, prepare for bitstream switching * / config c = getConfig (ConfigLen);

cutSamplesFlush =

FlushDecoder ();

ReConfigureDecoder (c);

/ * Get pre-mount AU and decode, discard output samples * / int numPreRollFrames = getNumPreRollFrames ();

for (auldx = 0; auldx <numPreRollFrames; auldx ++) int auLen = getAuLen ();

AU nextAU getPreRollAU (auLen);

0 DecodeAU (nextAU);

} / * Get regular AU and decode ★ /

AU au = UsacFrame ();

outSamplesGrame = Decode (au);

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/ * Apply crossfade * / for (i = 0; i <128; i ++) {outSamples [i] = outSamplesFlush [i] * (li / 127) + outSamplesFrame [i] * (i / 127)}

for (i = 128; i <outputFrameLength; i ++) {outSamples [i] = outSamplesFrame [i];

}} otherwise {go to error;

} }

}

Although some 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 stage or to a characteristic of a method stage. Similarly, the aspects described in the context of a method step also represent a description of a corresponding block or element or feature of a corresponding apparatus. Some or all of the method steps may be executed by (or using) a ti ολ «VMIUO» Λτφ x62

vi) AiAL ί ·

IrssTfT '/ Ώί THE hardware device, such as a microprocessor, a programmable computer, or an electronic circuit. In some embodiments, one or more of the most important method steps can be performed by such an apparatus. In the embodiments of the invention, the methods described herein are implemented by a processor or implemented by a computer.

Depending on certain implementation requirements, the embodiments of the invention can be implemented in hardware or software. The implementation can be done using a non-transient storage medium, such as a digital storage medium, for example, a floppy disk, a DVD, a Blu-Ray, a CD, a

ROM, PROM, EPROM, EEPROM, or FLASH memory, which has electronically readable control signals stored therein, which cooperate (or are capable of cooperating) with a programmable computer system in such a way as to carry out the respective method. Therefore, the digital storage medium can be computer readable.

Some embodiments in accordance with the invention comprise a data carrier having electronically readable control signals, which are capable of cooperating with a programmable computer system, such that one of the methods that is performed is carried out.

described in the present application.

In general, the embodiments of the present invention can be implemented as a computer program product with a program code, the program code is operative to carry out one of the methods when the computer program product is executed on a computer. . The program code can be stored, for example, in a machine-readable carrier.

Other modalities comprise the computer program for carrying out one of the methods described herein, stored in a machine-readable carrier.

In other words, one embodiment of the method of the invention, therefore, is a computer program that has program code to carry out one of the methods described herein, when the computer program is run on a computer. .

A further embodiment of the method of the invention, therefore, is a data carrier (or a digital storage medium, or a computer readable medium) comprising, recorded therein, the computer program for carrying out one of the methods described herein. The data carrier, the digital storage medium or the recorded medium are typically tangible and / or non-transient.

A further embodiment of the method of the invention, therefore, is a data stream or a sequence of signals representing the computer program for carrying out one of the methods described herein. The data stream or signal sequence, for example, can be configured to be transferred over a data communication connection, for example over the Internet.

A further embodiment comprises a processing means, eg, a computer, or a programmable, programmed, configured, or adapted logic device to carry out one of the methods described herein.

An additional embodiment comprises a computer having the computer program installed therein to carry out one of the methods described herein.

A further embodiment according to the invention comprises an apparatus or a system configured to transfer (for example, electronically or optically) a computer program for carrying out one of the methods described in this application, to a receiver. The receiver may be, for example, a computer, a mobile device, a memory device, or the like. The apparatus or system may comprise, for example, a

JNSTh'JTfj OE LA. · File server to transfer the computer program to the receiver.

In some embodiments, a programmable logic device (eg, an array of field programmable gates) may be used to perform some or all of the functionality of the methods described herein. In some embodiments, an array of field programmable gates can cooperate with a microprocessor to carry out one of the methods described herein. In general, the methods are preferably carried out by any hardware apparatus.

The embodiments described above are merely illustrative for the principles of the present invention. It is understood that modifications and variations of the arrangements and details described herein will be apparent to others with experience in the art. Therefore, it is intended to be limited only by the scope of the impending patent claims, and not by the specific details presented by way of description and explanation of the embodiments herein.

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Priority claims
Document	Office	Kind	Date
13189328	European Patent Office (EPO)	A	2013-10-18
13189328	European Patent Office (EPO)	A	2013-10-18
131893281	European Patent Office (EPO)	–	2013-10-18
2014072063	European Patent Office (EPO)	W	2014-10-14
2014072063	European Patent Office (EPO)	W	2014-10-14
131893281	–	–	–
EP20130189328	–	–	–
PCTEP2014072063	–	–	–
WO2014EP72063	–	–	–

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EP2863386A1	European Patent Office (EPO)	A1	2015-04-22
CA2925653A1	Canada	A1	2015-04-23
WO2015055683A1	World Intellectual Property Organization (WIPO)	A1	2015-04-23
TW201523587A	Taiwan Province of China	A	2015-06-16
AR098075A1	Argentina	A1	2016-04-27
AU2014336243A1	Australia	A1	2016-05-26
KR20160060686A	Republic of Korea	A	2016-05-30
SG11201602971SA	Singapore	A	2016-05-30
CN105745704A	China	A	2016-07-06
EP3044782A1	European Patent Office (EPO)	A1	2016-07-20
MX2016004845A	Mexico	A	2016-07-26
US2016232910A1	United States of America	A1	2016-08-11
JP2016539357A	Japan	A	2016-12-15
AU2014336243B2	Australia	B2	2017-02-02
TWI579832B	Taiwan Province of China	B	2017-04-21
BR112016008415A2	Brazil	A2	2017-08-22
EP3044782B1	European Patent Office (EPO)	B1	2017-09-06
JP6214765B2	Japan	B2	2017-10-18
RU2016118985A	Russian Federation	A	2017-11-23
ES2644370T3	Spain	T3	2017-11-28
ZA201603154B	South Africa	B	2017-11-29
PT3044782T	Portugal	T	2017-12-04
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MX355274BThis record	Mexico	B	2018-04-13
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US2025061906A1	United States of America	A1	2025-02-20

Audio decoder, apparatus for generating encoded audio output data and methods permitting initializing a decoder.

Abstract

Term

YES. ;

IMPI

FRAUNHOFER-GESELLSCHAFT FORSCHUNG EV

THE DIVISIONAL DIRECTOR OF PATENTS

NAHANNY CANAL REYES

MX / 2018/30364

INITIALIZATION OF A DECODER

FIELD OF THE INVENTION

BRIEF DESCRIPTION OF THE INVENTION

-C-

DETAILED DESCRIPTION OF THE INVENTION

I

INDUSTRIAL

INSTITUTE

IN;

OF THE C

INDUSTRIAL

I

TM

Priority claims9

Members55

Events

Numbers

Titles2

Classification