公开(公告)号：US20130282382A1

公开(公告)日：2013-10-24

申请号：US13901960

申请日：2013-05-24

Applicant: DOLBY INTERNATIONAL AB

Inventor： Per Hedelin ,  Pontus Carlsson ,  Leif Jonas Samuelsson ,  Michael Schug

IPC: G10L19/26

CPC classification number: G10L19/26 ,  G10L19/008 ,  G10L19/032 ,  G10L19/035

Abstract: The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.

Abstract translation: 本发明教导了一种新的音频编码系统，其可以以低比特率良好地对一般音频和语音信号进行编码。 所提出的音频编码系统包括用于基于自适应滤波器对输入信号进行滤波的线性预测单元; 变换单元，用于将经滤波的输入信号的帧变换为变换域; 以及用于量化变换域信号的量化单元。 量化单元基于输入信号特性来决定用基于模型的量化器或非基于模型的量化器对变换域信号进行编码。 优选地，该决定基于由变换单元应用的帧大小。

公开(公告)号：US12183355B2

公开(公告)日：2024-12-31

申请号：US18620081

申请日：2024-03-28

Applicant: Dolby Laboratories Licensing Corporation ,  DOLBY INTERNATIONAL AB

Inventor： Jeffrey Riedmiller ,  Harald Mundt ,  Michael Schug ,  Martin Wolters

IPC: G10L19/22 ,  G10L19/02 ,  G10L19/16 ,  G10L19/26 ,  H03G3/30 ,  H03G3/32 ,  H03G7/00

Abstract: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

公开(公告)号：US11670315B2

公开(公告)日：2023-06-06

申请号：US17750803

申请日：2022-05-23

Applicant: Dolby Laboratories Licensing Corporation ,  DOLBY INTERNATIONAL AB

Inventor： Jeffrey Riedmiller ,  Harald Mundt ,  Michael Schug ,  Martin Wolters

IPC: G10L19/22 ,  G10L19/16 ,  G10L19/02 ,  H03G7/00 ,  H03G3/32 ,  G10L19/26 ,  H03G3/30

CPC classification number: G10L19/22 ,  G10L19/02 ,  G10L19/0208 ,  G10L19/167 ,  G10L19/26 ,  H03G3/3089 ,  H03G3/32 ,  H03G7/007

Abstract: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

公开(公告)号：US11195536B2

公开(公告)日：2021-12-07

申请号：US16800553

申请日：2020-02-25

Applicant: DOLBY INTERNATIONAL AB

Inventor： Michael Schug ,  Holger Hoerich ,  Tobias R. Wagenblass ,  Christof Fersch ,  Karsten Linzmeier

IPC: G10L19/002 ,  G06K9/62

Abstract: The present document describes a method for allocating bits to a frame of a sequence of frames to yield a bitstream having a constant average bitrate, wherein the frame comprises audio data and metadata. The method comprises maintaining an overall bit reservoir and maintaining a virtual bit reservoir being a subset of the overall bit reservoir, such that bits for the metadata of the frame are allocated from the virtual bit reservoir and such that bits for the audio data of the frame are allocated from the overall bit reservoir.

公开(公告)号：US10573324B2

公开(公告)日：2020-02-25

申请号：US16079943

申请日：2017-02-01

Applicant: DOLBY INTERNATIONAL AB

Inventor： Michael Schug ,  Holger Hoerich ,  Tobias R. Wagenblass ,  Christof Fersch ,  Karsten Linzmeier

IPC: G10L19/002 ,  G06K9/62

Abstract: The present document describes a method (800) for allocating bits to a frame (301) of a sequence of frames (301) to yield a bitstream having a constant average bitrate, wherein the frame (301) comprises audio data and metadata. The method (800) comprises maintaining (801) an overall bit reservoir (100) and maintaining (802) a virtual bit reservoir (510) being a subset of the overall bit reservoir (100), such that bits for the metadata of the frame (301) are allocated from the virtual bit reservoir (510) and such that bits for the audio data of the frame (301) are allocated from the overall bit reservoir (100).

公开(公告)号：US10418045B2

公开(公告)日：2019-09-17

申请号：US15482328

申请日：2017-04-07

Applicant: DOLBY INTERNATIONAL AB ,  Dolby Laboratories Licensing Corporation

Inventor： Jeffrey Riedmiller ,  Harald Mundt ,  Michael Schug ,  Martin Wolters

IPC: G10L19/22 ,  G10L19/16 ,  H03G7/00 ,  G10L19/02 ,  H03G3/32 ,  G10L19/26 ,  H03G3/30

Abstract: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

公开(公告)号：US09697840B2

公开(公告)日：2017-07-04

申请号：US14359697

申请日：2012-11-28

Applicant: DOLBY INTERNATIONAL AB

Inventor： Arijit Biswas ,  Marco Fink ,  Michael Schug

IPC: G10L19/02 ,  G10L19/038 ,  G10L25/54 ,  G10H1/00 ,  G10H1/38 ,  G10L19/022 ,  G10L21/0388

CPC classification number: G10L19/02 ,  G10H1/0008 ,  G10H1/383 ,  G10H2210/066 ,  G10H2250/225 ,  G10L19/022 ,  G10L19/038 ,  G10L21/0388 ,  G10L25/54

Abstract: The present document relates to methods and systems for music information retrieval (MIR). In particular, the present document relates to methods and systems for extracting a chroma vector from an audio signal. A method (900) for determining a chroma vector (100) for a block of samples of an audio signal (301) is described. The method (900) comprises receiving (901) a corresponding block of frequency coefficients derived from the block of samples of the audio signal (301) from a core encoder (412) of a spectral band replication based audio encoder (410) adapted to generate an encoded bitstream (305) of the audio signal (301) from the block of frequency coefficients; and determining (904) the chroma vector (100) for the block of samples of the audio signal (301) based on the received block of frequency coefficients.

公开(公告)号：US20150269950A1

公开(公告)日：2015-09-24

申请号：US14439795

申请日：2013-11-11

Applicant: DOLBY INTERNATIONAL AB ,  DOLBY LABORATORIES LICENSING CORPORATION

Inventor： Michael Schug ,  Phillip Williams

IPC: G10L19/008 ,  G10L19/032 ,  G10L19/16 ,  G10L19/02

CPC classification number: G10L19/008 ,  G10L19/02 ,  G10L19/032 ,  G10L19/173

Abstract: The present document relates to audio encoding/decoding. In particular, the present document relates to a method and system for reducing the complexity of a bit allocation process used in the context of audio encoding/decoding. An audio encoder (300) configured to encode an audio signal according to a first audio codec system is described. The audio encoder (300) comprises a transform unit (302) configured to determine a set of spectral coefficients (312) based on the audio signal. Furthermore, the encoder (300) comprises a floating-point encoding unit (304) configured to determine a set of scale factors and a set of scaled values (314), based on the set of spectral coefficients (312); and to encode the set of scale factors to yield a set of encoded scale factors (313). In addition, the encoder (300) comprises a bit allocation and quantization unit (305, 306) configured to determine a total number of available bits for quantizing the set of scaled values (314), based on a first target data-rate and based on the number of bits used for the set of encoded scale factors (313); to determine a first control parameter (315) indicative of an allocation of the total number of available bits for quantizing the scaled values of the set of scaled values (314); and to quantize the set of scaled values (314) in accordance to the first control parameter (315) to yield a set of quantized scaled values (317). Furthermore, the encoder (300) comprises a transcoding simulation unit (320) configured to determine a second control parameter (321) based on the first control parameter (315); wherein the second control parameter (321) enables a transcoder to convert the first bitstream into a second bitstream at a second target data-rate; wherein the second bitstream accords to a second audio codec system different from the first audio codec system; and wherein the first bitstream comprises the second control parameter.

公开(公告)号：US08924201B2

公开(公告)日：2014-12-30

申请号：US13901960

申请日：2013-05-24

Applicant: Dolby International AB

Inventor： Per Hedelin ,  Pontus Carlsson ,  Leif Jonas Samuelsson ,  Michael Schug

IPC: G10L19/02 ,  G10L19/26 ,  G10L19/035 ,  G10L19/008

CPC classification number: G10L19/26 ,  G10L19/008 ,  G10L19/032 ,  G10L19/035

Abstract: The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.

Abstract translation: 本发明教导了一种新的音频编码系统，其可以以低比特率良好地对一般音频和语音信号进行编码。 所提出的音频编码系统包括用于基于自适应滤波器对输入信号进行滤波的线性预测单元; 变换单元，用于将经滤波的输入信号的帧变换为变换域; 以及用于量化变换域信号的量化单元。 量化单元基于输入信号特性来决定用基于模型的量化器或非基于模型的量化器对变换域信号进行编码。 优选地，该决定基于由变换单元应用的帧大小。

公开(公告)号：US20140188488A1

公开(公告)日：2014-07-03

申请号：US14184961

申请日：2014-02-20

Applicant: Dolby International AB ,  Dolby Laboratories Licensing Corporation

Inventor： Michael Schug ,  Phillip Williams ,  Luca Baradel

IPC: G10L19/008

CPC classification number: G10L19/008 ,  G10L19/02 ,  G10L19/032 ,  G10L19/173

Abstract: An audio encoder configured to encode an audio signal to generate a bitstream having E-AC-3 format, including by determining a first control parameter indicative of an allocation of available mantissa bits for quantized audio content of the signal. The encoder is configured to perform transcoding simulation to determine a second control parameter in a manner based at least in part on statistical analysis of results of E-AC-3 bit allocation processing of audio data assuming a first target data rate, and of AC-3 bit allocation processing of the data assuming a second target data rate, and to include the second control parameter in the bitstream for use by a converter to convert the bitstream into a second to bitstream having AC-3 format at the second target data rate. Other aspects are converters configured to perform transcoding on a bitstream using such a second control parameter, and methods performed by any embodiment of the inventive encoder or converter.

Abstract translation: 音频编码器被配置为对音频信号进行编码以生成具有E-AC-3格式的比特流，包括通过确定指示信号的量化音频内容的可用尾数位的分配的第一控制参数。 编码器被配置为执行代码转换模拟，以至少部分地基于假定第一目标数据速率的音频数据的E-AC-3位分配处理的结果的统计分析和AC- 假设第二目标数据速率的数据的3比特分配处理，并且将第二控制参数包括在比特流中以供转换器使用以将比特流转换为具有第二目标数据速率的具有AC-3格式的第二比特流。 其他方面是被配置为使用这样的第二控制参数在比特流上执行代码转换的转换器，以及由本发明的编码器或转换器的任何实施例执行的方法。