ТРАНСФОРМАЦИЯ ШКАЛЫ ВРЕМЕНИ КАДРОВ В ШИРОКОПОЛОСНОМ ВОКОДЕРЕ

Изобретение относится к трансформации шкалы времени, т.е. расширению или сжатию, кадров в вокодере и, в частности, к способам трансформации шкалы времени кадров в широкополосном вокодере. Техническим результатом является повышение качества трансформированных по шкале времени кадров и снижение вычислительной нагрузки. Указанный технический результат достигается тем, что способ передачи речи включает трансформацию шкалы времени остаточного низкополосного речевого сигнала в растянутую или сжатую версию остаточного низкополосного речевого сигнала, трансформацию шкалы времени высокополосного речевого сигнала в растянутую или сжатую версию высокополосного речевого сигнала и объединение подвергнутых трансформации шкалы времени низкополосного и высокополосного речевых сигналов для получения полного трансформированного по шкале времени речевого сигнала. Трансформация шкалы времени высокополосного речевого сигнала содержит определение множества периодов основного тона из низкополосного речевого сигнала ...

ОСНОВАННАЯ НА ЛИНЕЙНОМ ПРЕДСКАЗАНИИ СХЕМА КОДИРОВАНИЯ, ИСПОЛЬЗУЮЩАЯ ФОРМИРОВАНИЕ ШУМА В СПЕКТРАЛЬНОЙ ОБЛАСТИ

Изобретение относится к способу кодирования аудио сигнала и средствам для осуществления этого способа. Технический результат изобретения заключается в создании концепции кодирования, позволяющей уменьшить сложность при сопоставимой или даже увеличенной эффективности кодирования. Концепция кодирования, основанная на линейном предсказании при использовании спектрального разложения входного аудио сигнала для вычисления коэффициентов линейного предсказания, использует формирование шума в спектральной области на основании вычисленных коэффициентов линейного предсказания. Эффективность кодирования может сохраняться, даже если используется такое перекрывающееся преобразование для спектрального разложения, которое вызывает наложение и требует отмены наложения во времени, такое как критически дискретизированное перекрывающееся преобразование, например MDCT. 3 н. и 10 з.п. ф-лы, 4ил.

СПОСОБ И УСТРОЙСТВО ДЛЯ ОПРЕДЕЛЕНИЯ НАИМЕНЬШЕГО ЦЕЛОГО ЧИСЛА БИТОВ, ТРЕБУЕМОГО ДЛЯ ПРЕДСТАВЛЕНИЯ НЕДИФФЕРЕНЦИРУЕМЫХ ЗНАЧЕНИЙ КОЭФФИЦИЕНТОВ УСИЛЕНИЯ, ДЛЯ СЖАТИЯ ПРЕДСТАВЛЕНИЯ КАДРА ДАННЫХ HOA

Изобретение относится к средствам для определения наименьшего целого числа битов, требуемого для представления недифференцируемых значений коэффициентов усиления. Технический результат заключается в повышении точности определения требуемого числа битов. Каждый сигнал канала в каждом кадре содержит группу значений выборки. Каждому сигналу канала каждого из кадров данных HOA присваивают дифференцируемое значение коэффициента усиления. При этом дифференцируемое значение коэффициента усиления вызывает изменение амплитуд первых значений выборки сигнала канала в текущем кадре ((k-2)) данных HOA по отношению ко вторым значениям выборки сигнала канала в предыдущем кадре ((k-3)) данных HOA. При этом результирующие сигналы каналов с адаптированным коэффициентом усиления кодируют в кодирующем устройстве. При этом представление кадра данных HOA выполняют в пространственной области для O сигналов w(t) виртуальных громкоговорителей, при этом положения виртуальных громкоговорителей лежат на единичной ...

ГАРМОНИЧЕСКОЕ ПРЕОБРАЗОВАНИЕ, УСОВЕРШЕНСТВОВАННОЕ ПЕРЕКРЕСТНЫМ ПРОИЗВЕДЕНИЕМ

Изобретение относится к системам звукового кодирования, которые используют способ гармонического преобразования для высокочастотной реконструкции (HFR). Технический результат заключается в повышении качества кодируемого аудиосигнала. Система генерирования высокочастотной составляющей сигнала из низкочастотной составляющей сигнала включает блок анализирующих фильтров, создающий набор сигналов анализируемых поддиапазонов низкочастотной составляющей сигнала. Она также включает блок нелинейной обработки, предназначенный для генерирования сигнала синтезируемого поддиапазона с синтезируемой частотой путем модификации фазы первого и второго сигналов анализируемых поддиапазонов из набора сигналов анализируемых поддиапазонов и комбинирования сигналов анализируемых поддиапазонов с модифицированной фазой. Она также включает блок синтезирующих фильтров, предназначенный для генерирования высокочастотной составляющей сигнала из сигнала синтезируемого поддиапазона. 5 н. и 17 з.п. ф-лы, 30 ил.

ЗВУКОВОЕ КОДИРУЮЩЕЕ УСТРОЙСТВО И ДЕКОДЕР ДЛЯ КОДИРОВАНИЯ ДЕКОДИРОВАНИЯ ФРЕЙМОВ КВАНТОВАННОГО ЗВУКОВОГО СИГНАЛА

Звуковое кодирующее устройство (10), приспособленное для кодирования фреймов квантованного звукового сигнала для получения кодированных фреймов, где фрейм включает ряд звуковых образцов временной области. Звуковое кодирующее устройство (10) включает этап анализа предиктивного кодирования (12) для определения информации о коэффициентах синтезирующего фильтра и фрейма области предсказания, основанного на фрейме звуковых образцов. Звуковое кодирующее устройство (10) далее включает преобразователь, вводящий временное совмещение имен (14), для преобразования перекрывающихся фреймов области предсказания в частотную область для получения спектров фрейма области предсказания, где преобразователь, вводящий временное совмещение имен (14), приспособлен для преобразования перекрывающихся фреймов области предсказания способом критической выборки. Кроме того, звуковое кодирующее устройство (10) включает кодирующее устройство, уменьшающее избыточность (16) для кодирования спектров фрейма области предсказания ...

УСТРОЙСТВО ДЛЯ МИКШИРОВАНИЯ МНОЖЕСТВА ВХОДНЫХ ДАННЫХ

Изобретение относится к устройствам для микширования множества входных потоков данных для получения потока данных, которые могут применяться, например, в области систем конференц-связи, включая системы видео- и телеконференций. Техническим результатом является уменьшение сложности вычислений при микшировании кодированных с помощью SBR-кодера аудиосигналов. Указанный результат достигается тем, что устройство (500) для микширования первого фрейма (540-1) первого входного потока данных (510-1) и второго фрейма (540-2) второго входного потока данных (510-2) содержит блок обработки (520), предназначенный для формирования выходного фрейма (550), где выходной фрейм (550) содержит выходные спектральные данные, характеризующие нижнюю часть выходного спектра до выходной частоты перехода, и где выходной фрейм содержит выходные SBR-данные, характеризующие верхнюю часть выходного спектра выше выходной частоты перехода посредством значений энергии в выходном разрешении временно-частотной сетки; процессорный ...

КОДИРОВАНИЕ МНОГОКАНАЛЬНОГО АУДИО

Изобретение относится к устройствам кодирования многоканального аудио. Техническим результатом является создание устройства кодирования N аудио сигналов в М аудио сигналов и ассоциированные параметрические данные, N>M, M≥1, позволяющего повысить качество восстановленного многоканального сигнала. Указанный технический результат достигается за счет того, что устройство (10) кодирования многоканального аудио содержит первый и второй блоки кодирования (110, 120). Первый блок (110) кодирует многоканальный аудио сигнал (101) в пространственное сведение (102) и первые ассоциированные параметрические данные (104), обеспечивающие возможность многоканальному устройству (20) декодирования восстанавливать многоканальный аудио сигнал (203) из пространственного сведения (102). Второй блок (120) генерирует, исходя из пространственного сведения (102), вторые ассоциированные параметрические данные (105), которые обеспечивают возможность устройству декодирования восстанавливать пространственное сведение ...

ГАРМОНИЧЕСКОЕ ПРЕОБРАЗОВАНИЕ, УСОВЕРШЕНСТВОВАННОЕ ПЕРЕКРЕСТНЫМ ПРОИЗВЕДЕНИЕМ

Изобретение относится к системам звукового кодирования, которые используют способ гармонического преобразования для высокочастотной реконструкции (HFR). Описана система и способ генерирования высокочастотной составляющей сигнала из низкочастотной составляющей сигнала. Система включает блок анализирующих фильтров, создающий набор сигналов анализируемых поддиапазонов низкочастотной составляющей сигнала. Она также включает блок нелинейной обработки, предназначенный для генерирования сигнала синтезируемого поддиапазона с синтезируемой частотой путем модификации фазы первого и второго сигналов анализируемых поддиапазонов из набора сигналов анализируемых поддиапазонов и комбинирования сигналов анализируемых поддиапазонов с модифицированной фазой. В конечном счете, она включает блок синтезирующих фильтров, предназначенный для генерирования высокочастотной составляющей сигнала из сигнала синтезируемого поддиапазона. Технический результат - улучшение эффективности кодирования традиционных кодеков ...

СИСТЕМА И СПОСОБ ВОЗБУЖДЕНИЯ СМЕШАННОЙ КОДОВОЙ КНИГИ ДЛЯ КОДИРОВАНИЯ РЕЧИ

Изобретение относится к средствам возбуждения смешанной кодовой книги для кодирования речи. Технический результат заключается в повышении воспринимаемого качества речевого сигнала по сравнению с системами кодирования, использующими только импульсное возбуждение или только шумовое возбуждение. Способ кодирования аудио/речевого сигнала включает в себя определение вектора смешанной кодовой книги на основании поступающего аудио/речевого сигнала, причем вектор смешанной кодовой книги включает в себя сумму записи первой кодовой книги из первой кодовой книги и записи второй кодовой книги из второй кодовой книги. Способ дополнительно включает в себя генерацию кодированного аудиосигнала на основании определенного вектора смешанной кодовой книги и передачу индекса кодированного возбуждения определенного вектора смешанной кодовой книги. 3 н. и 23 з.п. ф-лы, 17 ил.

КОДИРОВАНИЕ И ДЕКОДИРОВАНИЕ АУДИОСИГНАЛОВ

Изобретение относится к средствам для кодирования/декодирования информации аудиосигнала. Технический результат заключается в повышении эффективности кодирования/декодирования. Считывают кодированную информацию аудиосигнала, содержащую: кодированное представление аудиосигнала для первого и второго кадров; первую информацию для первого кадра и первый элемент управляющих данных, имеющий первое значение; вторую информацию основного тона для второго кадра и второй элемент управляющих данных, имеющий второе значение, отличное от первого значения, причем первый и второй элементы управляющих данных находятся в одном и том же поле; и третий элемент управляющих данных для первого кадра, второго кадра и третьего кадра. Третий элемент управляющих данных указывает на наличие или отсутствие первой информации основного тона и/или второй информации основного тона. Третий элемент управляющих данных кодируется в одном бите, имеющем значение, которое отличает третий кадр от первого и второго кадров. Третий ...

ОСНОВАННОЕ НА ЛИНЕЙНОМ ПРЕДСКАЗАНИИ КОДИРОВАНИЕ АУДИО С ИСПОЛЬЗОВАНИЕМ УЛУЧШЕННОЙ ОЦЕНКИ РАСПРЕДЕЛЕНИЯ ВЕРОЯТНОСТЕЙ

... 1. Основанный на линейном предсказании аудиодекодер, содержащий:модуль (102) оценки распределений вероятностей, сконфигурированный с возможностью определять, для каждой из множества спектральных компонент, оценку (28) распределения вероятностей из информации коэффициентов линейного предсказания, содержащейся в потоке (22) данных, в который закодирован аудиосигнал;каскад (104) энтропийного декодирования и деквантования, сконфигурированный с возможностью осуществлять энтропийное декодирование и деквантование спектра (26), составленного из упомянутого множества спектральных компонент, из потока (22) данных с использованием оценки распределения вероятностей, которая определена для каждой из упомянутого множества спектральных компонент; ифильтр, сконфигурированный с возможностью формировать спектр (26) согласно передаточной функции, зависящей от синтезирующего фильтра линейного предсказания, определенного посредством информации коэффициентов линейного предсказания,при этом модуль оценки распределений ...

ЗАВИСЯЩЕЕ ОТ ЭНЕРГИИ КВАНТОВАНИЕ ДЛЯ ЭФФЕКТИВНОГО КОДИРОВАНИЯ ПРОСТРАНСТВЕННЫХ ПАРАМЕТРОВ ЗВУКА

... 1. Устройство квантования параметра, предназначенное для квантования входного параметра, причем входной параметр представляет собой меру, характеризующую одиночный канал или пару каналов относительно другого одиночного канала или пары каналов многоканального сигнала, содержащее: генератор правила квантования, предназначенный для генерации правила квантования на основании соотношения меры величины энергии канала или пары каналов и меры величины энергии многоканального сигнала; и устройство квантования значения, предназначенное для получения квантованного параметра из входного параметра с использованием сгенерированного правила квантования. 2. Устройство квантования параметра по п. 1, в котором генератор правила квантования функционирует таким образом, что осуществляет генерацию такого правила квантования, что квантование является более грубым для того канала или для той пары каналов, которые имеют низкую меру величины энергии, чем для того канала или для той пары каналов, которые имеют высокую ...

ГЕНЕРАЦИЯ КОМФОРТНОГО ШУМА

... 1. Способ генерирования параметров управления комфортного шума, CN, отличающийся тем, что:сохраняют (S1; 1a) CN-параметрыдля кадров дескриптора добавления тишины, SID, и активных кадров затягивания в буфере (200) заранее определенного размера (M);определяют (S2, 1b, 2) подмножество CN-параметров (Q,E), релевантное для SID-кадров, на основе возраста сохраненных CN-параметров и на основе остаточных энергий;используют (S3, 3, 4) определенное подмножество CN-параметров (Q,E) для определения параметров управления CN (q,E) для первого SID-кадра («Первого SID»), следующего за активным кадром сигнала.2. Способ по п. 1, отличающийся тем, что:обновляют (1a) для SID-кадров и активных кадров затягивания буфер (200) посредством новых CN-параметров;обновляют (1b) для активных кадров без затягивания, размера K подмножества (Q,E) с ограничением по возрасту сохраненных CN-параметров на основе числа pпоследовательных активных кадров без затягивания;выбирают (2) подмножество (Q,E) CN-параметров из подмножества ...

МАСШТАБИРУЕМЫЙ АУДИОКОДЕК БЕЗ ПОТЕРЬ И АВТОРСКОЕ ИНСТРУМЕНТАЛЬНОЕ СРЕДСТВО

... 1. Способ кодирования и авторского создания аудиоданных, содержащий этапы: кодируют без потерь аудиоданные в последовательности окон анализа для масштабируемого битового потока; сравнивают буферизированную полезную нагрузку для кодированных аудиоданных с разрешенной полезной нагрузкой для каждого окна и масштабируют кодированные без потерь аудиоданные в окнах, не соответствующих требованиям, так, чтобы буферизированная полезная нагрузка для битового потока не превышала разрешенную полезную нагрузку, при этом этап масштабирования вводит потери в кодированные данные в этих окнах. 2. Способ по п.1, в котором аудиоданные разделяют на части старших значащих разрядов (СтР) и младших значащих разрядов (МлР) для каждого окна анализа и кодируют с помощью различных алгоритмов без потерь. 3. Способ по п.2, в котором аудиоданные разделяют с помощью этапов: назначают минимальную битовую ширину СтР (минимальное количество СтР); вычисляют функцию стоимости для аудиоданных в окне анализа; если функция ...

ДЕКОДИРОВАНИЕ КОДИРОВАННЫХ С ПРЕДСКАЗАНИЕМ ДАННЫХ С ИСПОЛЬЗОВАНИЕМ АДАПТАЦИИ БУФЕРА

... 1. Способ, включающий: ! прием потока, содержащего кодированные данные и информацию предсказания, связанную с кодированными данными, при этом информация предсказания создана на основе данных в буфере кодирования с предсказанием; ! прием коэффициента, указывающего величину повышающей или понижающей дискретизации, которой требуется подвергнуть кодированные данные в качестве части процесса декодирования кодированных данных; ! формирование декодированных данных из кодированных данных с использованием принятых коэффициента и информации предсказания; ! буферизацию по меньшей мере части декодированных данных в одном или более буферов, при этом по меньшей мере один из одного или более буферов имеет по меньшей мере один размер, отличный от соответствующего размера буфера кодирования с предсказанием; ! идентификацию по меньшей мере части буферизованных декодированных данных для использования при декодировании последующих кодированных данных; и ! модифицирование идентифицированных данных для обеспечения ...

КОДИРОВАНИЕ МНОГОКАНАЛЬНОГО АУДИО

... 1. Устройство (10) кодирования многоканального аудио для кодирования N аудиосигналов (101) в М аудиосигналов (102) и ассоциированные параметрические данные (104, 105), где М и N являются целыми числами, N>М, M≥1, причем устройство (10) кодирования многоканального аудио содержит: ! первый блок (110) для кодирования N аудиосигналов (101) в М аудиосигналов (102) и первые ассоциированные параметрические данные (104), причем М аудиосигналов (102) и первые ассоциированные параметрические данные (104) представляют N аудиосигналов (101); и ! второй блок (120), соединенный с первым блоком (110), причем второй блок (120) выполнен с возможностью генерирования, исходя из М аудиосигналов (102), вторых ассоциированных параметрических данных (105), представляющих М аудиосигналов (102), причем вторые ассоциированные параметрические данные содержат параметры модифицирования, обеспечивающие возможность восстановления M аудиосигналов (102) из K дополнительных аудиосигналов (103), представляющих собой альтернативное ...

ДЕКОДИРОВАНИЕ КОДИРОВАННЫХ С ПРЕДСКАЗАНИЕМ ДАННЫХ С ИСПОЛЬЗОВАНИЕМ АДАПТАЦИИ БУФЕРА

... 3Изобретение относится к декодированию сжатой цифровой информации, в частности к декодированию битовых потоков, отражающих контент, который сжат с применением методов кодирования с долговременным предсказанием. Декодер (например, декодер AAC-LTP) принимает поток, содержащий кодированные аудиоданные и данные предсказания. Кодированные данные во время декодирования подвергают повышающей или понижающей дискретизации. Части декодированных данных хранят в буфере для их использования при декодировании последующих кодированных данных. Буфер, в который помещают декодированные данные, имеет размеры, отличные от размеров буфера, используемого в кодере при создании кодированных данных. Часть данных в буфере декодера идентифицируют и модифицируют с использованием перемежающихся нулевых значений, чтобы обеспечить соответствие размерам буфера кодирования с предсказанием в кодере. Технический результат - повышение качества звука при декодировании. 6 н. и 13 з.п. ф-лы, 11 ил.

УСТРОЙСТВО КОДИРОВАНИЯ АУДИО, СПОСОБ КОДИРОВАНИЯ АУДИО, ПРОГРАММА КОДИРОВАНИЯ АУДИО, УСТРОЙСТВО ДЕКОДИРОВАНИЯ АУДИО, СПОСОБ ДЕКОДИРОВАНИЯ АУДИО И ПРОГРАММА ДЕКОДИРОВАНИЯ АУДИО

Изобретение относится к средствам для маскирования ошибок при кодировании/декодировании аудио. Технический результат заключается в восстановлении качества звучания без увеличения алгоритмической задержки, когда происходит потеря пакета при кодировании аудио. Устройство передачи аудиосигнала для кодирования аудиосигнала включает в себя блок кодирования аудио, который кодирует аудиосигнал, и блок кодирования побочной информации, который вычисляет и кодирует побочную информацию из прогнозного сигнала. Устройство приема аудиосигнала для декодирования аудиокода и вывода аудиосигнала включает в себя буфер аудиокода, который обнаруживает потерю пакета на основе состояния приема аудиопакета, блок декодирования аудиопараметров, который декодирует аудиокод, когда аудиопакет принят корректно, блок декодирования побочной информации, который декодирует код побочной информации, когда аудиопакет принят корректно, блок сбора побочной информации, который собирает побочную информацию, получаемую посредством ...

СПОСОБ И УСТРОЙСТВО ДЛЯ ОПРЕДЕЛЕНИЯ НАИМЕНЬШЕГО ЦЕЛОГО ЧИСЛА БИТОВ, ТРЕБУЕМОГО ДЛЯ ПРЕДСТАВЛЕНИЯ НЕДИФФЕРЕНЦИРУЕМЫХ ЗНАЧЕНИЙ КОЭФФИЦИЕНТОВ УСИЛЕНИЯ, ДЛЯ СЖАТИЯ ПРЕДСТАВЛЕНИЯ КАДРА ДАННЫХ HOA

ПОДДИАПАЗОННЫЙ РЕЧЕВОЙ КОДЕКС С МНОГОКАСКАДНЫМИ ТАБЛИЦАМИ КОДИРОВАНИЯ И ИЗБЫТОЧНЫМ КОДИРОВАНИЕМ

... 1. Способ, содержащий этапы, на которых: ! в средстве обработки аудио обрабатывают поток битов для аудиосигнала, при этом поток битов содержит: ! основную кодированную информацию для текущего кадра, который ссылается на сегмент предыдущего кадра, который должен быть использован в декодировании текущего кадра; и ! избыточную кодированную информацию для декодирования текущего кадра, причем избыточная кодированная информация содержит информацию предыстории сигнала, связанного с указанным ссылкой сегментом предыдущего кадра; и ! выводят результат. ! 2. Способ по п.1, в котором средством обработки аудио является речевой кодер реального времени, а результатом является кодированная речь. ! 3. Способ по п.1, в котором информация предыстории сигнала содержит предысторию возбуждения для указанного ссылкой сегмента, но не предысторию возбуждения для одного или более не указанных ссылкой сегментов предыдущего кадра. ! 4. Способ по п.1, в котором средством обработки аудио является речевой декодер, и ...

ГЕНЕРАЦИЯ СИГНАЛА ВЕРХНЕЙ ПОЛОСЫ

АУДИОКОДЕР И СПОСОБ ДЛЯ КОДИРОВАНИЯ АУДИОСИГНАЛА

АУДИОКОДЕК, ИСПОЛЬЗУЮЩИЙ СИНТЕЗ ШУМА В ТЕЧЕНИЕ НЕАКТИВНОЙ ФАЗЫ

... 1. Аудиокодер, содержащий:- модуль (12) оценки фонового шума, выполненный с возможностью непрерывно обновлять параметрическую оценку фонового шума в течение активной фазы (24) на основе входного аудиосигнала;- кодер (14) для кодирования входного аудиосигнала в поток данных в течение активной фазы; и- детектор (16), выполненный с возможностью обнаруживать вход в неактивную фазу (28) после активной фазы (24) на основе входного аудиосигнала,- при этом аудиокодер выполнен с возможностью, при обнаружении входа в неактивную фазу, кодировать в поток данных параметрическую оценку фонового шума, непрерывно обновляемую в течение активной фазы, после которой следует обнаруженная неактивная фаза.2. Аудиокодер по п. 1, в котором модуль (12) оценки фонового шума выполнен с возможностью, при непрерывном обновлении параметрической оценки фонового шума, осуществлять различение между компонентом шума и компонентом полезного сигнала во входном аудиосигнале и определять параметрическую оценку фонового шума ...

УСТРОЙСТВО И СПОСОБ ДЛЯ МАСКИРОВАНИЯ ОШИБОК ПРИ СТАНДАРТИЗИРОВАННОМ КОДИРОВАНИИ РЕЧИ И АУДИО С НИЗКОЙ ЗАДЕРЖКОЙ (USAC)

... 1. Устройство (100) для формирования спектральных замещающих значений для аудиосигнала, содержащее:- буферный блок (110) для сохранения предыдущих спектральных значений, связанных с ранее принимаемым безошибочным аудиокадром, и- формирователь (120) кадров маскирования для формирования спектральных замещающих значений, когда текущий аудиокадр не принят или является ошибочным, при этом ранее принимаемый безошибочный аудиокадр содержит информацию фильтра, причем информация фильтра имеет ассоциированное значение стабильности фильтра, указывающее стабильность прогнозного фильтра, и при этом формирователь (120) кадров маскирования выполнен с возможностью формировать спектральные замещающие значения на основе предыдущих спектральных значений и на основе значения стабильности фильтра.2. Устройство (100) по п. 1, в котором формирователь (120) кадров маскирования выполнен с возможностью формировать спектральные замещающие значения посредством произвольной смены знака предыдущих спектральных значений ...

Verfahren und Vorrichtung zur Glättung von Grundperiodewellenformen

CODEBUCHSTRUKTUR UND SUCHVERFAHREN FÜR DIE SPRACHKODIERUNG

VORRICHTUNG UND VERFAHREN ZUR ERZEUGUNG EINES PEGELPARAMETERS UND VORRICHTUNG UND VERFAHREN ZUR ERZEUGUNG EINER MEHRKANAL-REPRÄSENTATION

Noise output for a decoded speech signal

Noise output for a decoded speech signal

A speech decoder (201) is arranged and constructed to receive a plurality of speech parameters and decode the plurality of speech parameters into at least one fragment of decoded speech. A noise generator (205) is arranged and constructed to output a noise signal. Determining means (203) provide a decision as to whether the plurality of speech parameters represents unvoiced speech. A switch (209), operating together with the speech decoder, the noise generator, and the means for determining, is arranged and constructed to output the noise signal when the plurality of speech parameters represents unvoiced speech. A noise source (2051) e.g. a pseudo-random sequence generator applies a Gaussian noise signal to an LPC filter (2053) which spectrally shapes the noise signal with an estimate of the original speech spectrum envelope as provided by the LPC parameters provided by speech decoder (201). A filter (2055) adjusts the output energy of the noise signal to match the original decoded speech ...

Speech coding

Voice quality testing of digital wireless networks in particular tetra networks using identical sound cards

Voice quality testing of digital wireless networks (such as TETRA) is carried out using a pair of units 29, 30, each unit 29, 30 having a substantially identical sound card 20. Each unit 29, 30 has a port 24 for connection to a communication terminal 32, 34 allowing audio signals to be transmitted over the wireless network by one unit 29 or 30 and received by the other 30 or 29. One unit 29 or 30 derives the audio signal to be transmitted from an originating data set and the other unit 30 or 29 converts the audio signal to produce a received data set. The received data set is then analyzed against the originating data set to determine the voice quality of the network. The invention eliminates the effects of degradation in sampling the signal by having substantially identical first and second sound cards of same type, quality and manufacture and for the cards to play and record at the same quality in order to achieve full standardization for the analysis.

A DIGITAL SPEECH TRANSMISSION SYSTEM

Audio coding method and apparatus

Generation of comfort noise

Generation of comfort noise

PROCEDURE AND DEVICE FOR PRÄDIKTIVEN THE QUANTIZATION OF VOTING CLINGING SPRACHSIGNALEN

SPRACHUBERTRAGUNGSSYSTEM AND PROCEDURE FOR THE TREATMENT OF LOSING DATA FRAMEWORKS

DEVICE FOR MIXING SEVERAL INPUT DATA STREAM

ENERGY-DEPENDENT QUANTIZATION FOR EFFICIENT CODING OF SPATIAL AUDIO PARAMETERS

MORE PRÄDIKTIONSSPRACHKODIERER WITH MUSTERAUSWAHL FOR CODING PATTERN FOR THE REDUCTION OF SENSITIVITY FOR FRAMEWORK ERRORS

SCALE-CASH LOSS-LOOSE AUDIO CODECS AND PRODUCTION TOOL

CODE BOOK STRUCTURE AND SEARCH METHOD FOR THE LANGUAGE CODING

CODING OF VOTING LOTS LANGUAGE SEGMENTS WITH LOW DATA RATE

Mechanism to the speech synthesis according to the principle of the impulse-excited Kanalvocoders

Audio signal encoding and decoding

An audio codec suitable for robust wireless transmission of high quality audio with low latency, still at a moderate bit rate. The encoding and decoding methods are based on ADPCM and in addition to the encoded output bits APM, additional data QB are included in output data blocks, namely data QB representing an internal value of the adaptive quantization ADQ of the ADPCM encoding algorithm, especially a scaling factor encoded and truncated to such as 8 bits. Further, output data blocks preferably include data CFB representing an internal value of the predictor PR of the ADPCM encoding algorithm, especially data CFB representing coefficients of a lattice prediction FIR filter which, truncated to such as 8 bits, can be sequentially included in output data blocks. These additional data QB, CFB regarding internal values of the ADPCM encoding algorithm can be utilized at the encoder side to increase robustness against loss of data blocks in wireless transmission. Especially, the decoding algorithm ...

Audio coding device, audio coding method, audio coding program, audio decoding device, audio decoding method, and audio decoding program

AUDIO CODING DEVICE, AUDIO CODING METHOD, AUDIO CODING PROGRAM, AUDIO DECODING DEVICE, AUDIO DECODING METHOD, AND AUDIO DECODING PROGRAM Disclosed is an audio encoding method and device. The audio encoding method is performed by an audio encoding device for encoding an audio signal, which comprises: an audio encoding step of encoding an audio signal; and a side information encoding step of calculating, from a look-ahead signal, side information for calculating a predicted value of an audio parameter for synthesizing a decoded audio, and encoding the side information. The audio encoding device for encoding an audio signal comprises: an audio encoding unit that encodes an audio signal; and a side information encoding unit that calculates, from a look-ahead signal, side information for calculating a predicted value of an audio parameter for synthesizing a decoded audio, and encodes the side information.

Generation of a multichannel encoded signal and decoding of a multichannel encoded signal

Lossless multi-channel audio codec using adaptive segmentation with random access point (RAP) and multiple prediction parameter set (MPPS) capability

A lossless audio codec encodes/decodes a lossless variable bit rate (VBR) bitstream with random access point (RAP) capability to initiate loss less decoding at a specified segment within a frame and/or multiple prediction parameter set (MPPS) capability partitioned to mitigate transient effects. This is accomplished with an adaptive segmentation technique that fixes segment start points based on constraints imposed by the existence of a desired RAP and/or detected transient in the frame and selects a optimum segment duration in each frame to reduce encoded frame payload subject to an encoded segment payload constraint RAP and MPPS are particularly applicable to improve overall performance for longer frame durations.

Mixing of input data streams and generation of an output data stream therefrom

Parameter decoding device, parameter encoding device, and parameter decoding method

Codebook structure and search for speech coding

Injection high frequency noise into pulse excitation for low bit rate celp

Apparatus and method for coding a portion of an audio signal using a transient detection and a quality result

An apparatus for coding a portion of an audio signal (10) to obtain an encoded audio signal (26) for the portion of the audio signal comprises a transient detector (12) for detecting whether a transient signal is located in the portion of the audio signal to obtain a transient detection result (14), an encoder stage (16) for performing a first encoding algorithm on the audio signal, the first encoding algorithm having a first characteristic, and for performing a second encoding algorithm on the audio signal, the second encoding algorithm having a second characteristic being different from the first characteristic, a processor (18) for determining which encoding algorithm results in an encoded audio signal being a better approximation to the portion of the audio signal with respect to the other encoding algorithm to obtain a quality result (20), and a controller (22) for determining whether the encoded audio signal for the portion of the audio signal is to be generated by either the first ...

Linear prediction coefficient generation during frame erasure or packet loss

Audio decoder and method for providing a decoded audio information using an error concealment based on a time domain excitation signal

Abstract An audio decoder for providing a decoded audio information on the basis of an encoded audio information, the audio decoder comprising: an error concealment configured to 5 provide an error concealment audio information for concealing a loss of an audio frame following an audio frame encoded in a frequency domain representation using a time domain excitation signal; wherein the error concealment is configured to modify a time domain excitation signal obtained on the basis of one or more audio frames preceding a lost audio frame, in order to obtain the error concealment audio information; wherein the 10 error concealment is configured to time-scale the time domain excitation signal obtained on the basis of one or more audio frames preceding a lost audio frame, or the one or more copies thereof, in dependence on a prediction of a pitch for the time of the one or more lost audio frames. 9697917 1 (GHMatters) P102970.AU.4 ...

LOW BIT-RATE PATTERN ENCODING AND DECODING CAPABLE OF REDUCING AN INFORMATION TRANSMISSION RATE

CONVERSION APPARATUS AND CONVERSION METHOD OF SPEECH CODE SEQUENCE

A voice code sequence converting device and method for converting a code sequence with low computational complexity by receiving a first code sequence having a pitch period at an input terminal on the input side, converting the first code sequence into a second code sequence having a pitch period, and outputting the second code sequence from an output terminal on the output side. In addition to a circuit for synthesizing a decoded signal from a code sequence of the CELP method on the input side, the voice code sequence converting device has a circuit for directly delivering the LP coefficient and pitch period decoded by an LP coefficient decoding circuit (12) and a pitch component decoding circuit (13) respectively to an LP coefficient encoding circuit (31) and a pitch component calculating circuit (40) on the output side respectively so as to deliver them to code sequence conversion of the output side. Therefore, the LP analysis of the decoded signal by the output side and the selection ...

A SIGNAL MODIFICATION METHOD FOR EFFICIENT CODING OF SPEECH SIGNALS

SPEECH CODEC EMPLOYING SPEECH CLASSIFICATION FOR NOISE COMPENSATION

A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. For each bit rate mode selected, pluralities of fixed or innovation subcodebooks are selected for use in generating innovation vectors. The speech coder distinguishes various voice signals as a function of their voice content. For example, a Voice Activity Detection (VAD) algorithm selects an appropriate coding scheme depending on whether the speech signal comprises active or inactive speech. The encoder may consider varying characteristics of the speech signal including sharpness, a delay correlation, a zero-crossing rate, and a residual energy. In another embodiment of the present invention, code excited linear ...

LOSSLESS MULTI-CHANNEL AUDIO CODEC USING ADAPTIVE SEGMENTATION WITH RANDOM ACCESS POINT (RAP) AND MULTIPLE PREDICTION PARAMETER SET (MPPS) CAPABILITY

A lossless audio codec encodes/decodes a lossless variable bit rate (VBR) bitstream with random access point (RAP) capability to initiate loss less decoding at a specified segment within a frame and/or multiple prediction parameter set (MPPS) capability partitioned to mitigate transient effects. This is accomplished with an adaptive segmentation technique that fixes segment start points based on constraints imposed by the existence of a desired RAP and/or detected transient in the frame and selects a optimum segment duration in each frame to reduce encoded frame payload subject to an encoded segment payload constraint RAP and MPPS are particularly applicable to improve overall performance for longer frame durations.

MULTI-MODE AUDIO CODEC AND CELP CODING ADAPTED THEREFORE

In accordance with a first aspect of the present invention, bitstream elements of sub-frames are encoded differentially to a global gain value so that a change of the global gain value of the frames results in an adjustment of an output level of the decoded representation of the audio content. Concurrently, the differential coding saves bits otherwise occurring when introducing a new syntax element into an encoded bitstream. Even further, the differential coding enables the lowering of the burden of globally adjusting the gain of an encoded bitstream by allowing the time resolution in setting the global gain value to be lower than the time resolution at which the afore-mentioned bitstream element differentially encoded to the global gain value adjusts the gain of the respective sub-frame. In accordance with another aspect, a global gain control across CELP coded frames and transform coded frames is achieved by co-controlling the gain of the codebook excitation of the CELP codec, along with ...

HIGH RESOLUTION AUDIO CODING

Methods, systems, and apparatus, including computer programs encoded on computer storage media, for performing long-term prediction (LTP) are described. One example of the methods includes determining a pitch gain and a pitch lag of an input audio signal for at least a predetermined number of frames. It is determined that the pitch gain of the input audio signal has exceeded a predetermined threshold and that a change of the pitch lag of the input audio signal has been within a predetermined range for at least the predetermined number of frames. In response to determining that a pitch gain of the input audio signal has exceeded the predetermined threshold and that the change of the third pitch lag has been within the predetermined range for at least the predetermined number of frames, a pitch gain is set for a current frame of the input audio signal.

ENCODER AND METHOD FOR ENCODING AN AUDIO SIGNAL WITH REDUCED BACKGROUND NOISE USING LINEAR PREDICTIVE CODING

It is shown an encoder for encoding an audio signal with reduced background noise using linear predictive coding. The encoder comprises a background noise estimator configured to estimate background noise of the audio signal, a background noise reducer configured to generate background noise reduced audio signal by subtracting the estimated background noise of the audio signal from the audio signal, and a predictor configured to subject the audio signal to linear prediction analysis to obtain a first set of linear prediction filter (LPC) coefficients and to subject the background noise reduced audio signal to linear prediction analysis to obtain a second set of linear prediction filter (LPC) coefficients. Furthermore, the encoder comprises an analysis filter composed of a cascade of time-domain filters controlled by the obtained first set of LPC coefficients and the obtained second set of LPC coefficients.

VOICE SYNTHESIS UTILIZING MULTI-LEVEL FILTER EXCITATION

CONVERSION APPARATUS AND CONVERSION METHOD OF SPEECH CODE SEQUENCE

A voice code sequence converting device and method for converting a code sequence with low computational complexity by receiving a first code sequence having a pitch period at an input terminal on the input side, converting the first code sequence into a second code sequence having a pitch period, and outputting the second code sequence from an output terminal on the output side. In addition to a circuit for synthesizing a decoded signal from a code sequence of the CELP method on the input side, the voice code sequence converting device has a circuit for directly delivering the LP coefficient and pitch period decoded by an LP coefficient decoding circuit (12) and a pitch component decoding circuit (13) respectively to an LP coefficient encoding circuit (31) and a pitch component calculating circuit (40) on the output side respectively so as to deliver them to code sequence conversion of the output side. Therefore, the LP analysis of the decoded signal by the output side and the selection ...

CONCEPT FOR ENCODING AN AUDIO SIGNAL AND DECODING AN AUDIO SIGNAL USING SPEECH RELATED SPECTRAL SHAPING INFORMATION

According to an aspect of the present invention an encoder for encoding an audio signal comprises an analyzer configured for deriving prediction coefficients and a residual signal from a frame of the audio signal. The encoder comprises a formant information calculator configured for calculating a speech related spectral shaping information from the prediction coefficients, a gain parameter calculator configured for calculating a gain parameter from an unvoiced residual signal and the spectral shaping information and a bitstream former configured for forming an output signal based on an information related to a voiced signal frame, the gain parameter or a quantized gain parameter and the prediction coefficients.

METHOD, APPARATUS, DEVICE, COMPUTER-READABLE MEDIUM FOR BANDWIDTH EXTENSION OF AN AUDIO SIGNAL USING A SCALED HIGH-BAND EXCITATION

A method includes determining a first modeled high-band signal based on a low-band excitation signal of an audio signal, where the audio signal includes a high-band portion and a low-band portion. The method also includes determining scaling factors based on energy of sub-frames of the first modeled high-band signal and energy of corresponding sub-frames of the high-band portion of the audio signal. The method includes applying the scaling factors to a modeled high-band excitation signal to determine a scaled high-band excitation signal and determining a second modeled high-band signal based on the scaled high-band excitation signal. The method includes determining gain parameters based on the second modeled high-band signal and the high-band portion of the audio signal.

AUDIO DECODER AND METHOD FOR PROVIDING A DECODED AUDIO INFORMATION USING AN ERROR CONCEALMENT MODIFYING A TIME DOMAIN EXCITATION SIGNAL

An audio decoder (200; 400) for providing a decoded audio information (220; 412) on the basis of an encoded audio information (210;410). The audio decoder comprises an error concealment (240; 480; 600) configured to provide an error concealment audio information (242;482;612) for concealing a loss of an audio frame, wherein the error concealment is configured to modify a time domain excitation signal (452,456;610) obtained for one or more audio frames preceding a lost audio frame, in order to obtain the error concealment audio information.

AUDIO DECODER AND METHOD FOR PROVIDING A DECODED AUDIO INFORMATION USING AN ERROR CONCEALMENT BASED ON A TIME DOMAIN EXCITATION SIGNAL

An audio decoder (100; 300) for providing a decoded audio information (112;312) on the basis of an encoded audio information (110; 310) comprises an error concealment (130; 380; 500) configured to provide an error concealment audio information (132;382;512) for concealing a loss of an audio frame following an audio frame encoded in a frequency domain representation (322) using a time domain excitation signal (532).

CROSS PRODUCT ENHANCED HARMONIC TRANSPOSITION

The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.

METHOD AND DEVICE FOR SPEECH SIGNAL PITCH PERIOD ESTIMATION AND CLASSIFICATION IN DIGITAL SPEECH CODERS

A method and a device for speech signal digital coding are provided, in which at each frame there is carried out a long-term analysis for estimating a pitch period 'd', a long-term prediction coefficient 'b', a gain 'G', and an apriori classification of the signal as active/inactive and, for an active signal, as voiced/unvoiced. Period estimation circuits compute the period on the basis of a suitably-weighted covariance function, and classification circuits distinguish voiced signals from unvoiced signals by comparing the long-term prediction coefficient and gain with frame-by-frame variable thresholds.

METHOD OF ADAPTING THE NOISE MASKING LEVEL IN AN ANALYSIS-BY-SYNTHESIS SPEECH CODER EMPLOYING A SHORT-TERM PERCEPTUAL WEIGHTING FILTER

In an analysis-by-synthesis speech coder employing a short-term perceptual weighting filter with transfer function W(z)=A(z/.gamma.1)/A(z/.gamma.2), the values of the spectral expansion coefficients .gamma.1 and .gamma.2 are adapted dynamically on the basis of spectral parameters obtained during short-term linear prediction analysis. The spectral parameters serving in this adaptation may in particular comprise parameters representative of the overall slope of the spectrum of the speech signal, and parameters representative of the resonant character of the short-term synthesis filter.

SPEECH PITCH LAG CODING APPARATUS AND METHOD

A pitch lag is extracted for each of a predetermined number of sub-frames. A predicted pitch lag for a pertinent sub- frame in the predetermined number of sub-frames is calculated on the basis of at least two pitch lags extracted for sub-frames other than the pertinent sub-frame or at least one pitch lag extracted for sub-frame other than the pertinent sub-frame and the preceding sub-frame by one sub-frame. A difference between the predicted pitch lag and the extracted pitch lag is then coded. Thus, an input speech signal pitch lag is coded for each sub-frame having a predetermined length.

SPEECH ENCODING METHOD AND SPEECH ENCODING SYSTEM

In this speech encoding system, the limiter circuit is input with the delay of adaptive codebook obtained for the previous subframe, and the pitch cycle search range is limited so that the delay of adaptive codebook obtained for the previous subframe is not discontinuous to the delay of adaptive codebook to be obtained for the current subframe, and the pitch cycle search range limited is output to the pitch calculation circuit. The pitch calculation circuit is input with output signal X w (n) of the perceptual weighting circuit and the pitch cycle search range output from the limiter, calculating the pitch cycle T op, then outputting at least one pitch cycle T op to the adaptive codebook circuit. The adaptive codebook circuit is input with the perceptual weighting signal x'w(n), the past excitation signal v(n) output from the gain quantization circuit, the perceptual weighting impulse response h w (n) output from the impulse response calculation circuit, and the pitch cycle T op from the ...

Audio signal encoding and decoding

Low-frequency emphasis for CPL-based coding in frequency domain

Method for reconstructing sub-band turbidity sound parameters at voice decoding end based on support vector machine

NUMERICAL VOCODEUR

METHOD AND APPARATUS FOR PLAYBACK OF A HIGHER-ORDER AMBISONICS AUDIO SIGNAL

IMPROVING NON-SPEECH CONTENT FOR LOW RATE CELP DECODER

오디오 신호 인코딩/디코딩 방법 및 오디오 신호 인코딩/디코딩 장치

... 본 발명의 실시예는 오디오 신호 인코딩 및 디코딩 방법, 오디오 신호 인코딩 및 디코딩 장치, 전송기, 수신기, 및 통신 시스템을 제공하며, 이것은 인코딩 및/또는 디코딩 성능을 향상시킬 수 있다. 오디오 신호 인코딩 방법은, 인코딩될 시간 도메인 신호를 저대역 신호 및 고대역 신호로 분할하는 단계; 상기 저대역 신호를 인코딩하여 저주파 인코딩 파라미터를 획득하는 단계; 상기 저주파 인코딩 파라미터에 따라 음성 정도 인자를 계산하고, 상기 저주파 인코딩 파라미터에 따라 고대역 여기 신호(high band excitation signal)를 예측하는 단계 - 상기 음성 정도 인자는 고대역 신호에 의해 제공되는 음성 특성(voiced characteristic)의 정도를 나타내는 데 사용됨 - ; 상기 음성 정도 인자를 사용함으로써 고대역 여기 신호 및 랜덤 노이즈를 가중하여 합성 여기 신호를 획득하는 단계; 및 상기 합성 여기 신호 및 상기 고대역 신호에 기초하여 고주파 인코딩 파라미터를 획득하는 단계를 포함한다. 본 발명의 실시예에서의 기술적 솔루션은 인코딩 또는 디코딩 효과를 향상시킬 수 있다.

CODING MODEL SELECTION

The invention relates to an encoder (200) comprising an input (201) for inputting frames of an audio signal, a LTP analysis block (209) for performing a LTP analysis to the frames of the audio signal to form LTP parameters on the basis of the properties of the audio signal, and at least a first excitation block (206) for performing a first excitation for frames of the audio signal, and a second excitation block (207) for performing a second excitation for frames of the audio signal. The encoder (200) further comprises a parameter analysis block (202) for analysing said LTP parameters, and an excitation selection block (203) for selecting one excitation block among said first excitation block (206) and said second excitation block (207) for performing the excitation for the frames of the audio signal on the basis of the parameter analysis. The invention also relates to a device, a system, a method, a module and a computer program product. © KIPO & WIPO 2007 ...

INFORMATION SIGNAL REPRESENTATION USING LAPPED TRANSFORM

AUDIO ENCODER AND DECODER WITH PROGRAM INFORMATION OR SUBSTREAM STRUCTURE METADATA

codificação de sinal de banda alta com o uso de múltiplas sub-bandas

Apparatus and method for processing a decoded audio signal in a spectral domain

An apparatus for processing a decoded audio signal (100) comprising a filter (102) for filtering the decoded audio signal to obtain a filtered audio signal (104), a time-spectral converter stage (106) for converting the decoded audio signal and the filtered audio signal into corresponding spectral representations, each spectral representation having a plurality of subband signals, a weighter (108) for performing a frequency selective weighting of the filtered audio signal by a multiplying subband signals by respective weighting coefficients to obtain a weighted filtered audio signal, a subtractor (112) for performing a subband-wise subtraction between the weighted filtered audio signal and the spectral representation of the decoded audio signal, and a spectral-time converter (114) for converting the result audio signal or a signal derived from the result audio signal into a time domain representation to obtain a processed decoded audio signal (116).

AUDIO CODING DEVICE, AUDIO CODING METHOD, AUDIO CODING PROGRAM, AUDIO DECODING DEVICE, AUDIO DECODING METHOD, AND AUDIO DECODING PROGRAM

AUDIO PROCESSOR AND METHOD FOR GENERATING A FREQUENCY ENHANCED AUDIO SIGNAL USING PULSE PROCESSING

PARAMETRIC AUDIO CODING

The invention provides coding (11) of an at least two-channel audio signal (L,R) by determining common frequencies (fcom) in the at least two channels (L,R) of the audio signal, which common frequencies occur in at least two of the at least two channels of the audio signal, and by representing respective sinusoidal components in respective channels at a given common frequency by a representation of the given common frequency (fcom) and a representation of respective amplitudes (A,ΔA) of the respective sinusoidal components at the given common frequency.

METHOD AND DEVICE FOR CODING LAG PARAMETER AND CODE BOOK PREPARING METHOD

A lag parameter coding means (215b) generates a code corresponding to a lag parameter value by using a lag parameter code book (215a). On the decoding side, a lag parameter value corresponding to the lag parameter code generated on the coding side is decoded by using the same lag parameter code book (215a) and outputted. In the lag parameter code book (215a), the relation between the lag parameter value and the corresponding code (P) is shown. The relation is so determined as to increase the rate at which the decoded lag parameter value of when a bit error occurs in the code deviates to approximately an integral multiple (including one time) or an integral submultiple of the decoded lag parameter value of when no bit error occurs. As a result, the auditory degradation of quality of decoded sound is suppressed even when the code has a bit error.

METHOD, MEDIUM, AND SYSTEM ENCODING/DECODING MULTI-CHANNEL SIGNAL

A multi-channel signal decoding method is provided. A down-mixed signal representative of a multi-channel signal is decoded, and parameters representing characteristic relations between channels of the multi-channel signal are decoded. An additional parameter is estimated by using the decoded parameters, and the decoded down-mixed signal is up-mixed by using the decoded parameters and the estimated parameter so as to decode the multi-channel signal.

Linear predictive residual representation via non-iterative spectral reconstruction

Method of encoding speech at medium to high bit rates while maintaining very high speech quality, as specifically directed to the coding of the linear predictive (LPC) residual signal using either its Fourier Transform magnitude or phase. In particular, the LPC residual of the speech signal is coded using minimum phase spectral reconstruction techniques by transforming the LPC residual signal in a manner approximately a minimum phase signal, and then applying spectral reconstruction techniques for representing the LPC residual signal by either its Fourier Transform magnitude or phase. The non-iterative spectral reconstruction technique is based upon cepstral coefficients through which the magnitude and phase of a minimum phase signal are related. The LPC residual as reconstructed and regenerated is used as an excitation signal to a LPC synthesis filter in the generation of analog speech signals via speech synthesis from which audible speech may be produced.

Method and an apparatus of decoding an audio signal

The present invention includes an audio signal receiving unit receiving the audio signal having a plurality of channel signals including an ambient component signal and a source component signal; an ambient component signal extracting unit extracting the ambient component signal of each of the channels based on correlation between the channel signals; an ambient component signal modifying unit modifying the ambient component signal using surround effect information; a source component signal extracting unit extracting the source component signal of each of the channels based on the correlation between the channel signals; a first signal output unit outputting the modified ambient component signal and the source component signal; and a second signal output unit outputting the audio signal or the source component signal. Accordingly, in an apparatus for decoding an audio signal and method thereof according to the present invention, an ambient component signal is extracted and modified based ...

Method and apparatus for encoding audio signal, and method and apparatus for decoding audio signal

Methods and apparatuses for encoding and decoding of an audio signal using a mixture of a time-frequency method and a parametric method according to the audio band are provided. An encoding method of an audio signal includes: dividing input audio signals into a plurality of audio bands; selecting a coding method for each audio band; encoding each audio band according to the selected coding method for each band; and generating a bit stream including all the data encoded for each audio band, wherein selecting a coding method for each band comprises selecting smaller encoded data either from a parametric coding method or a time frequency coding method.

Method and apparatus for searching fixed codebook

A method and apparatus for searching fixed codebook are provided. The method includes: obtaining a basic codebook which comprises position information of N pulses on M tracks, wherein N and M are positive integers; choosing n pulses as search pulses, wherein the n pulses are parts of the N pulses and n is a positive integer smaller than N; and replacing position information of the n search pulses respectively with other position information on the tracks to obtain a searched codebook; executing the search process for K times, wherein K is a positive integer larger than or equal to 2, at least two or more search pulses are chosen in one of the K search processes , and the chosen search pulses vary in each of the K search processes; and obtaining an optimal codebook from the basic codebook and the searched codebook according to a preset criterion.

Method of detecting a predetermined frequency band in an audio data signal, detection device and computer program corresponding thereto

A method is provided for detecting a predetermined frequency band in an audio data signal which has previously been coded according to a succession of data blocks, among which at least certain blocks contain respectively at least one set of spectral parameters representing a linear prediction filter. Such a method of detection implements, for a current block among the at least certain blocks and for which at least a plurality of spectral parameters of the set have been previously decoded, acts of: determining, among the plurality of previously decoded spectral parameters, the index of the first spectral parameter closest to a threshold frequency; calculating at least one criterion on the basis of the determined index; and deciding whether the predetermined frequency band is detected in the current block, as a function of the criterion calculated.

AUDIO DECODER AND METHOD FOR PROVIDING A DECODED AUDIO INFORMATION USING AN ERROR CONCEALMENT BASED ON A TIME DOMAIN EXCITATION SIGNAL

An audio decoder for providing a decoded audio information on the basis of an encoded audio information includes an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame following an audio frame encoded in a frequency domain representation using a time domain excitation signal.

Cross product enhanced harmonic transposition

The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.

Apparatus and method for generating a plurality of parametric audio streams and apparatus and method for generating a plurality of loudspeaker signals

An apparatus for generating a plurality of parametric audio streams from an input spatial audio signal obtained from a recording in a recording space has a segmentor and a generator. The segmentor is configured for providing at least two input segmental audio signals from the input spatial audio signal, wherein the at least two input segmental audio signals are associated with corresponding segments of the recording space. The generator is configured for generating a parametric audio stream for each of the at least two input segmental audio signals to obtain the plurality of parametric audio streams.

Coding of audio signals

An encoder comprises an input for inputting frames of an audio signal in a frequency band, an analysis filter dividing the frequency band into lower and higher frequency bands, a first encoding block for encoding the audio signals of the lower frequency band, a second encoding block for encoding the audio signals of the higher frequency band, and a mode selector for selecting an operating mode for the encoder among at least a first mode where signals only on the lower frequency band are encoded, and a second mode where signals on both the lower and higher frequency band are encoded. The encoder has a scaler to gradually change the encoding properties of the second encoding block in connection with a change in the operating mode of the encoder. The invention also relates to a device, a decoder, a method, a module, a computer program product, and a signal.

Hearing aid with audio codec and method

A hearing aid comprising a time domain codec. The codec comprises a decoder adapted to generate a decoded output signal based on an input quantization index and an encoder for generating an output quantization index based on an input signal, said encoder comprising said decoder and a predictor receiving an excitation signal derived from said decoder output signal and outputting a prediction signal. The output quantization index is determined by repeated decoding of the quantization indices in order to minimize the error between the input signal and the prediction signal, and the predictor uses a recursive autocorrelation estimate for the error minimization. The invention further provides a method of encoding an audio signal.

Device and method for quantizing the gains of the adaptive and fixed contributions of the excitation in a celp codec

A device and method for quantizing a gain of a fixed contribution of an excitation in a frame, including sub-frames, of a coded sound signal. The gain is estimated in a sub-frame using a frame classification parameter, and is then quantized in the sub-frame using the estimated gain. The device and method can be used in jointly quantizing gains of adaptive and fixed contributions of an excitation. For retrieving a quantized gain of a fixed contribution of an excitation in a sub-frame, the gain of the fixed excitation contribution is estimated using a frame classification parameter, a gain codebook supplies a correction factor in response to a received, gain codebook index, and a multiplier multiplies the estimated gain by the correction factor to provide the quantized gain.

Multi-mode audio codec and celp coding adapted therefore

In an embodiment, bitstream elements of sub-frames are encoded differentially to a global gain value so that a change of the global gain value results in an adjustment of an output level of the decoded representation of the audio content. Concurrently, the differential coding saves bits. Even further, the differential coding enables the lowering of the burden of globally adjusting the gain of an encoded bitstream. In another embodiment, a global gain control across CELP coded frames and transform coded frames is achieved by co-controlling the gain of the codebook excitation of the CELP codec, along with a level of the transform or inverse transform of the transform coded frames. In another embodiment, the gain value determination in CELP coding is performed in the weighted domain of the excitation signal.

Information signal representation using lapped transform

An information signal reconstructor is configured to reconstruct, using aliasing cancellation, an information signal from a lapped transform representation of the information signal including, for each of consecutive, overlapping regions of the information signal, a transform of a windowed version of the respective region, wherein the information signal reconstructor is configured to reconstruct the information signal at a sample rate which changes at a border between a preceding region and a succeeding region of the information signal.

Coding device, coding method, decoding device, decoding method, and storage medium

For respective sampling data of waveform data of sounds to be coded, a prediction residual value is calculated as sampling residual data, and an effective bit length is calculated from this residual waveform data. Then, for the effective bit length data, a maximum effective bit length among processing targets is generated as common effective actual data, and coded data in which this common effective actual data and information indicating the common effective bit length are arranged in a predetermined configuration format are generated. The information included in the coded data is analyzed and each of the plurality of the common effective bit information is extracted. Then, waveform data of the sounds are decoded by performing inverse linear prediction processing from an analysis result on the residual waveform data decompressed by performing bit extension which adds a portion other than the common effective bit length.

Method and apparatus for encoding and decoding high frequency for bandwidth extension

Disclosed are a method and apparatus for encoding and decoding a high frequency for bandwidth extension. The method includes: estimating a weight; and generating a high frequency excitation signal by applying the weight between random noise and a decoded low frequency spectrum.

Apparatus and method for encoding and decoding an audio signal using an aligned look-ahead portion

An apparatus for encoding an audio signal having a stream of audio samples has: a windower for applying a prediction coding analysis window to the stream of audio samples to obtain windowed data for a prediction analysis and for applying a transform coding analysis window to the stream of audio samples to obtain windowed data for a transform analysis, wherein the transform coding analysis window is associated with audio samples within a current frame of audio samples and with audio samples of a predefined portion of a future frame of audio samples being a transform-coding look-ahead portion, wherein the prediction coding analysis window is associated with at least the portion of the audio samples of the current frame and with audio samples of a predefined portion of the future frame being a prediction coding look-ahead portion, wherein the transform coding look-ahead portion and the prediction coding look-ahead portion are identically to each other or are different from each other by less than 20%; and an encoding processor for generating prediction coded data or for generating transform coded data.

Apparatus and method for error concealment in low-delay unified speech and audio coding

An apparatus for generating spectral replacement values for an audio signal has a buffer unit for storing previous spectral values relating to a previously received error-free audio frame. Moreover, the apparatus includes a concealment frame generator for generating the spectral replacement values, when a current audio frame has not been received or is erroneous. The previously received error-free audio frame has filter information, the filter information having associated a filter stability value indicating a stability of a prediction filter. The concealment frame generator is adapted to generate the spectral replacement values based on the previous spectral values and based on the filter stability value.

Audio codec using noise synthesis during inactive phases

A parametric background noise estimate is continuously updated during an active or non-silence phase so that the noise generation may immediately be started with upon the entrance of an inactive phase following the active phase. In accordance with another aspect, a spectral domain is very efficiently used in order to parameterize the background noise thereby yielding a background noise synthesis which is more realistic and thus leads to a more transparent active to inactive phase switching.

Post-Quantization Gain Correction in Audio Coding

A gain adjustment apparatus for use in decoding of audio that has been encoded with separate gain and shape representations includes an accuracy meter configured to estimate an accuracy measure of the shape representation, and to determine a gain correction based on the estimated accuracy measure. An envelope adjuster further included in the apparatus is configured to adjust the gain representation based on the determined gain correction. 1. A method of operation by a gain adjustment apparatus , the method comprising:receiving an encoded audio signal comprising a set of gain values and a corresponding set of shape vectors, each gain value representing the energy of a frequency sub-band in a frequency transform of an input audio signal, and each corresponding shape vector representing a fine structure of the frequency transform in the frequency sub-band;determining an accuracy measure for each shape vector from corresponding shape quantization characteristics indicating a quantization resolution;determining a gain correction for each gain value as a function of the accuracy measure calculated for the corresponding shape vector; andadjusting each gain value according to the corresponding gain correction, to obtain corrected gain values, for use in decoding the encoded audio signal.2. The method of claim 1 , wherein each shape vector comprises a pulse vector and wherein determining the accuracy measure for the shape vector comprises calculating the accuracy measure as a function of the number of pulses allocated to the pulse vector claim 1 , as said quantization resolution claim 1 , and a maximum pulse height for the pulse vector claim 1 , and wherein greater pulse allocations correspond to higher accuracy and smaller pulse allocations correspond to lower accuracy.3. The method of claim 2 , further comprising determining the accuracy measure for each shape vector as a further function of the number of pulses allocated to the pulse vector in relation to a bandwidth of ...

Coding Method, Decoding Method, Coder, and Decoder

A coding method, a decoding method, a coder, and a decoder, where the coding method includes obtaining the pulse distribution, on a track, of the pulses to be encoded on the track, determining a distribution identifier for identifying the pulse distribution according to the pulse distribution, and generating a coding index that includes the distribution identifier. The decoding method includes receiving a coding index, obtaining a distribution identifier from the coding index, wherein the distribution identifier is configured to identify the pulse distribution, on a track, of the pulses to be encoded on the track, determining the pulse distribution, on a track, of all the pulses to be encoded on the track according to the distribution identifier, and reconstructing the pulse order on the track according to the pulse distribution. 2. The coder of claim 1 , wherein the instructions further cause the processor to be configured to:obtain pulse sign information indicating positive and negative features of the pulses; anddetermine a pulse sign index corresponding to the pulse sign information, and the coding index further comprising the pulse sign index corresponding to the pulse sign information for each pulse.3. The coder of claim 1 , wherein the instructions further cause the processor to be configured to:overlay information about the second and third indices with the first index used as a first value, a value of the first index corresponding to an independent value range of the coding index, using, in the overlaid information, a combination of the second index and the third index according to I3×W(N)+I2, the I2 representing the second index, the I3 representing the third index, and the W(N) representing a total quantity of all possible distributions of the pulse positions on the track.5. The method of claim 4 , further comprising:obtaining, by the coder, pulse sign information indicating positive and negative features of each pulse; anddetermining, by the coder, a ...

PACKET LOSS CONCEALMENT FOR SPEECH CODING

A speech coding method of reducing error propagation due to voice packet loss, is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame, the excitation of a next frame is obtained according to the reduced or limited pitch gain value of the first subframe, and the next frame is encoded according to the obtained excitation. The method is used for a voiced speech class. 1. A method for encoding an audio signal , wherein the audio signal is encoded frame-by-frame by an encoder , and each frame comprises a plurality of subframes , the method comprising:for a current frame that is to be encoded, obtaining an excitation of the current frame according to a reduced or limited pitch gain value of a first subframe of a previous frame, wherein the current frame is successive to the previous frame, and wherein the reduced or limited pitch gain value of the first subframe of the previous frame is obtained by reducing or limiting an initial pitch gain value of the first subframe of the previous frame; andencoding the current frame of the digital audio signal according to the excitation of the current frame.2. The method of claim 1 , wherein the reduced or limited pitch gain value of the first subframe of the previous frame is smaller than the initial pitch gain value of the first subframe claim 1 , and wherein reducing or limiting the initial pitch gain value of the first subframe to obtain the reduced or limited pitch gain value of the first subframe comprises:multiplying a scaling factor to the initial pitch gain value of the first subframe to obtain the reduced or limited pitch gain value of the first subframe,wherein the scaling factor is smaller than 1 and greater than 0.3. The method of claim 1 , wherein the reduced or limited pitch gain value of the first subframe is smaller than 1.4. The method of claim 1 , further comprising:inputting the excitation of the current frame to a Linear Prediction or Short- ...

Method of encoding, method of decoding, encoder, and decoder of an audio signal

The invention concerns an audio signal encoding method comprising the steps of: collecting the audio signal samples, determining sinusoidal components in subsequent frames, estimation of amplitudes and frequencies of the components for each frame, merging thus obtained pairs into sinusoidal trajectories, splitting particular trajectories into segments, transforming particular trajectories to the frequency domain by means of a digital transform performed on segments longer than the frame duration, quantization and selection of transform coefficients in the segments, entropy encoding, and outputting the quantized coefficients as output data. The method is characterized in that the length of the segments into which each trajectory is split is individually adjusted in time for each trajectory.

OPTIMIZED SCALE FACTOR FOR FREQUENCY BAND EXTENSION IN AN AUDIO FREQUENCY SIGNAL DECODER

A method and device are provided for determining an optimized scale factor to be applied to an excitation signal or a filter during a process for frequency band extension of an audio frequency signal. The band extension process includes decoding or extracting, in a first frequency band, an excitation signal and parameters of the first frequency band including coefficients of a linear prediction filter, generating an excitation signal extending over at least one second frequency band, filtering using a linear prediction filter for the second frequency band. The determination method includes determining an additional linear prediction filter, of a lower order than that of the linear prediction filter of the first frequency band, the coefficients of the additional filter being obtained from the parameters decoded or extracted from the first frequency and calculating the optimized scale factor as a function of at least the coefficients of the additional filter. 1. A method for determining an optimized scale factor to be applied to an excitation signal or to a filter in a method of extending a frequency band of an audio frequency signal ,the method comprising steps of:computing of a frequency response, R, of a linear prediction filter of a frequency band,{'sub': 'smoothed', 'smoothing of the value of R, so as to obtain R, the smoothing method being selected, from a group of smoothing methods including at least two smoothing methods, in function of a set of parameters comprising a plurality of parameters including the value of spectral slope, tilt, wherein the set of smoothing methods comprises an exponential smoothing with a factor being fixed over time.'}2. The method of claim 1 , wherein the exponential smoothing is of the type:{'br': None, 'i': R', 'R', 'R, 'sub': smoothed', 'precomputed', 'prev, '=0.5 +0.5 ,'}{'sub': prev', 'smoothed', 'precomputed, 'where Rcorresponds to the value of Rin the previous subframe, Rcorresponds to the value of R as computed during the ...

Optimized scale factor for frequency band extension in an audio frequency signal decoder

A method and device are provided for determining an optimized scale factor to be applied to an excitation signal or a filter during a process for frequency band extension of an audio frequency signal. The band extension process includes decoding or extracting, in a first frequency band, an excitation signal and parameters of the first frequency band including coefficients of a linear prediction filter, generating an excitation signal extending over at least one second frequency band, filtering using a linear prediction filter for the second frequency band. The determination method includes determining an additional linear prediction filter, of a lower order than that of the linear prediction filter of the first frequency band, the coefficients of the additional filter being obtained from the parameters decoded or extracted from the first frequency and calculating the optimized scale factor as a function of at least the coefficients of the additional filter.

AUDIO SIGNAL ENCODING AND DECODING METHOD, AND AUDIO SIGNAL ENCODING AND DECODING APPARATUS

An audio signal encoding and decoding method, an audio signal encoding and decoding apparatus, a transmitter, a receiver, and a communications system, which can improve encoding and/or decoding performance. The audio signal encoding method includes dividing a to-be-encoded time domain signal into a low band signal and a high band signal; encoding the low band signal to obtain a low frequency encoding parameter; calculating a voiced degree factor, and predicting a high band excitation signal; weighting the high band excitation signal and random noise using the voiced degree factor, so as to obtain a synthesized excitation signal; and obtaining a high frequency encoding parameter based on the synthesized excitation signal and the high band signal. Technical solutions in the embodiments of the present invention can improve an encoding or decoding effect. 1. An audio signal encoding method , comprising:encoding a low band signal of an audio signal to obtain one or more low frequency encoding parameters;calculating a voiced factor according to the one or more low frequency encoding parameters;obtaining a high band excitation signal according to the one or more low frequency encoding parameters;obtaining an emphasized noise by performing an emphasis operation on a random noise;obtaining a synthesized excitation signal according to the emphasized noise, the high band excitation signal and the voiced factor; andobtaining a high frequency encoding parameter based on the synthesized excitation signal and a high band signal of the audio signal.2. The method according to claim 1 , wherein obtaining a synthesized excitation signal according to the emphasized noise claim 1 , the high band excitation signal and the voiced factor comprises:obtaining the synthesized excitation signal by calculating a weighted sum of the high band excitation signal and the emphasized noise;wherein weighting factor for calculating the weighted sum is obtained according to the voiced factor.3. The ...

METHOD AND APPARATUS FOR HIGH FREQUENCY DECODING FOR BANDWIDTH EXTENSION

Disclosed are a method and an apparatus for high frequency decoding for bandwidth extension. The method for high frequency decoding for bandwidth extension comprises the steps of: decoding an excitation class; transforming a decoded low frequency spectrum on the basis of the excitation class; and generating a high frequency excitation spectrum on the basis of the transformed low frequency spectrum. The method and apparatus for high frequency decoding for bandwidth extension according to an embodiment can transform a restored low frequency spectrum and generate a high frequency excitation spectrum, thereby improving the restored sound quality without an excessive increase in complexity. 1. A high frequency decoding method comprising:decoding a low frequency spectrum and an excitation class for a current frame;modifying the low frequency spectrum by applying a random sign or an original sign to the low frequency spectrum based on the excitation class; andgenerating a high frequency excitation spectrum based on the modified low frequency spectrum.2. The high frequency decoding method of claim 1 , wherein the excitation class indicates one among a plurality of classes including a speech excitation class claim 1 , a first non-speech excitation class claim 1 , and a second non-speech excitation class.3. The high frequency decoding method of claim 2 , wherein the first non-speech excitation class is related to noisy characteristic and the second non-speech excitation class is related to tonal characteristic.4. The high frequency decoding method of claim 1 , wherein the modifying of the low frequency spectrum further comprises:normalizing the low frequency spectrum;identifying a control parameter based on the decoded excitation class; andmodifying the normalized low frequency spectrum by reducing an amplitude of the normalized low frequency spectrum based on the control parameter.5. The high frequency decoding method of claim 4 , wherein an amount of the reduced amplitude ...

Audio decoder for audio channel reconstruction

A method and apparatus for reconstructing N audio channels from M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal representing the M audio channels and decoding the encoded audio signal to obtain a frequency domain representation of the M audio channels. The method further includes extracting a parameter from the bitstream and reconstructing at least one of the N audio channels using the parameter. The parameter represents an angle between two signals, at least one of which is included in the M audio channels.

Audio Decoder Having A Bandwidth Extension Module With An Energy Adjusting Module

An audio decoder configured to produce an audio signal from a bitstream containing audio frames includes: a core band decoding module configured to derive a directly decoded core band audio signal from the bitstream; a bandwidth extension module configured to derive a parametrically decoded bandwidth extension audio signal from the core band audio signal and from the bitstream, wherein the bandwidth extension audio signal is based on a frequency domain signal having at least one frequency band; and a combiner configured to combine the core band audio signal and the bandwidth extension audio signal so as to produce the audio signal; wherein the bandwidth extension module includes an energy adjusting module being configured in such way that in a current audio frame in which an audio frame loss occurs, an adjusted signal energy for the current audio frame for the at least one frequency band is set. 11. Audio decoder configured to produce an audio signal (AS) from a bitstream (BS) containing audio frames (AF) , the audio decoder () comprising:{'b': '2', 'a core band decoding module () configured to derive a directly decoded core band audio signal (CBS) from the bitstream (BS);'}{'b': '3', 'a bandwidth extension module () configured to derive a parametrically decoded bandwidth extension audio signal (BES) from the core band audio signal (CBS) and from the bitstream (BS), wherein the bandwidth extension audio signal (BES) is based on a frequency domain signal (FDS) having at least one frequency band (FB); and'}{'b': '4', 'a combiner () configured to combine the core band audio signal (CBS) and the bandwidth extension audio signal (BES) so as to produce the audio signal (AS);'}{'b': 3', '5', '2', '2, 'wherein the bandwidth extension module () comprises an energy adjusting module () being configured in such way that in a current audio frame (AF) in which an audio frame loss (AFL) occurs, an adjusted signal energy for the current audio frame (AF) for the at least one ...

SYSTEMS AND METHODS FOR ENCODING AN AUDIO SIGNAL USING CUSTOM PSYCHOACOUSTIC MODELS

Systems and methods are provided for modifying an audio signal using custom psychoacoustic methods, for encoding the audio signal. A user's hearing profile is first obtained. Subsequently, a sample of the audio signal is split into frequency components. Next, masking and hearing thresholds are obtained from the user's hearing profile and applied to the frequency components of the audio sample, wherein the user's perceived data is calculated. User's imperceptible audio signal data is then disregarded. The audio sample is quantized and the resulting transformed audio sample encoded. 1. A method for modifying an audio signal for encoding the audio signal , the method comprising:obtaining a hearing profile;splitting a sample of the audio signal into frequency acomponents;obtaining masking thresholds from the hearing profile;obtaining hearing thresholds from the hearing profile;applying the masking and hearing thresholds to the frequency components and disregarding an imperceptible audio signal data;quantizing the audio signal; andencoding the audio signal.2. The method according to claim 1 , wherein the hearing profile is derived from at least one of a suprathreshold test claim 1 , a psychophysical tuning curve claim 1 , a threshold test and an audiogram.3. The method according to claim 1 , wherein the hearing profile is estimated from demographic information.4. The method according to claim 1 , wherein the hearing profile is derived from a psychophysical tuning curve and an audiogram.5. The method according to claim 4 , wherein the audiogram is derived from the psychophysical tuning curve.6. The method according to claim 1 , wherein the masking thresholds and hearing thresholds are applied to the frequency components of the audio signal and perceptual relevant information is calculated for the audio signal that is perceptually relevant.7. The method according to claim 6 , wherein perceptually relevant information is calculated by calculating perceptual entropy.8. The ...

Reuse of syntax element indicating vector quantization codebook used in compressing vectors

In general, techniques are described for indicating reuse of a syntax element indicating a vector quantization codebook used in compressing a vector. A device comprising a processor and a memory may perform the techniques. The processor may be configured to obtain a bitstream comprising a vector in a spherical harmonics domain. The bitstream may further comprise an indicator for whether to reuse, from a previous frame, a syntax element indicative of a vector quantization codebook used when compressing the vector. The memory may be configured to store the bitstream.

AUDIO ENCODER AND METHOD FOR ENCODING AN AUDIO SIGNAL

An audio encoder for providing an encoded representation on the basis of an audio signal, wherein the audio encoder is configured to obtain a noise information describing a noise included in the audio signal, and wherein the audio encoder is configured to adaptively encode the audio signal in dependence on the noise information, such that encoding accuracy is higher for parts of the audio signal that are less affected by the noise included in the audio signal than for parts of the audio signal that are more affected by the noise included in the audio signal. 1. An audio encoder for providing an encoded representation on the basis of an audio signal , wherein the audio encoder is configured to acquire a noise information describing a noise comprised by the audio signal , and wherein the audio encoder is configured to adaptively encode the audio signal in dependence on the noise information , such that encoding accuracy is higher for parts of the audio signal that are less affected by the noise comprised by the audio signal than for parts of the audio signal that are more affected by the noise comprised by the audio signal;wherein frequency components that are less corrupted by the noise are quantized with less error whereas components which are likely to comprise errors from the noise comprise a lower weight in the quantization process.2. The audio encoder according to claim 1 , wherein the audio encoder is configured to adaptively encode the audio signal by adjusting a perceptual objective function used for encoding the audio signal in dependence on the noise information.3. The audio encoder according to claim 1 , wherein the audio encoder is configured to simultaneously encode the audio signal and reduce the noise in the encoded representation of the audio signal claim 1 , by adaptively encoding the audio signal in dependence on the noise information.4. The audio encoder according to claim 1 , wherein the noise information is a signal-to-noise ratio.5. The audio ...

Device And Method For Quantizing The Gains Of The Adaptive And Fixed Contributions Of The Excitation In A Celp Codec

A device is for retrieving a quantized gain of a fixed contribution of an excitation in a sub-frame of a frame. The device includes a receiver of a gain codebook index; an estimator of the gain of the fixed contribution of the excitation in the sub-frame, wherein the estimator is supplied with a parameter representative of a classification of the frame; a gain codebook for supplying a correction factor in response to the gain codebook index; and a multiplier of the estimated gain by the correction factor to provide a quantized gain of the fixed contribution of the excitation in said sub-frame. 117-. (canceled)18. A device for retrieving a quantized gain of a fixed contribution of an excitation in a sub-frame of a frame , comprising:a receiver of a gain codebook index;an estimator of the gain of the fixed contribution of the excitation in the sub-frame, wherein the estimator is supplied with a parameter representative of a classification of the frame;a gain codebook for supplying a correction factor in response to the gain codebook index; anda multiplier of the estimated gain by the correction factor to provide a quantized gain of the fixed contribution of the excitation in said sub-frame.19. The device for retrieving the quantized gain of the fixed contribution of the excitation according to claim 18 , wherein the estimator comprises claim 18 , for a first sub-frame of the frame claim 18 , a calculator of a first estimation of the gain of the fixed contribution of the excitation in response to the parameter representative of the classification of the frame claim 18 , and a subtractor of an energy of a filtered innovation codevector from a fixed codebook from the first estimation to obtain the estimated gain.20. The device for retrieving the quantized gain of the fixed contribution of the excitation according to claim 18 , wherein the estimator claim 18 , for each sub-frame of said frame following the first sub-frame claim 18 , is responsive to the parameter ...

METHOD, APPARATUS AND SYSTEM FOR HYBRID SPEECH SYNTHESIS

A method of decoding an original speech signal for hybrid adversarial-parametric speech synthesis comprising: (a) receiving quantized original linear prediction coding parameters estimated by applying linear prediction coding analysis filtering to an original speech signal and a quantized compressed representation of a residual of the original speech signal; (b) dequantizing the original linear prediction coding parameters and the compressed representation of the residual; (c) inputting the dequantized compressed representation of the residual into a decoder part of a Generator for applying adversarial mapping from the compressed residual domain to a fake (first) signal domain; (d) outputting, by the decoder part of the Generator, a fake speech signal; (e) applying linear prediction coding analysis filtering to the fake speech signal for obtaining a corresponding fake residual; (f) reconstructing the original speech signal by applying linear prediction coding cross-synthesis filtering to the fake residual and the dequantized original linear prediction coding analysis parameters. 113-. (canceled)14. A method of decoding an original speech signal for hybrid adversarial-parametric speech synthesis , wherein the method includes the steps of:(a) receiving quantized original linear prediction coding parameters estimated by applying linear prediction coding analysis filtering to an original speech signal and a quantized compressed representation of a residual of the original speech signal;(b) dequantizing the original linear prediction coding parameters and the compressed representation of the residual;(c) inputting the dequantized compressed representation of the residual into a decoder part of a Generator for applying adversarial mapping from the compressed residual domain to a fake (first) signal domain;(d) outputting, by the decoder part of the Generator, a fake speech signal;(e) applying linear prediction coding analysis filtering to the fake speech signal for ...

SYSTEM AND METHOD FOR REDUCING TANDEMING EFFECTS IN A COMMUNICATION SYSTEM

The present disclosure is directed towards a system and method for reducing tandeming effects in a communications system. The method may include receiving, at a speech decoder, an input bitstream associated with an incoming initial speech signal from a speech encoder. The method may further include determining whether or not coding is required and if coding is required, modifying an excitation signal associated with the bitstream. The method may also include providing the modified excitation signal to an adaptive encoder. 1. A computer-implemented method for reducing tandeming effects in a communications system comprising:receiving, at a speech decoder, an input bitstream associated with an incoming initial speech signal from a speech encoder;determining whether or not coding is required;if coding is required, modifying an excitation signal associated with the bitstream; andproviding the modified excitation signal to an adaptive encoder.2. The method of claim 1 , further comprising:decoding T-milliseconds codec frames of the input bitstream that was encoded by a code excited linear prediction (“CELP”) based encoder at a rate of S kilobits/second for an adaptive encoder.3. The method of claim 1 , further comprising:calculating a total excitation signal u(n) by adding an adaptive and a fixed codebook vector, each scaled by a respective gain.4. The method of claim 3 , further comprising:saving the total excitation signal u(n) at an adaptive encoder memory, wherein at least one of the adaptive codebook vector and the fixed codebook vector include speech frames and discontinuous transmission (“DTX”) frames based on a CELP-based standard.5. The method of claim 2 , wherein the adaptive encoder includes an adaptive encoder memory configured to store a defined data structure.6. The method of claim 1 , further comprising:{'b': '1', 'i': 'n', 'calculating a new excitation signal e() for a new speech signal s(n) after VQA processing has been performed.'}7. The method of claim 6 ...

Method for reduction of aliasing introduced by spectral envelope adjustment in real-valued filterbanks

The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

SIGNAL RE-USE DURING BANDWIDTH TRANSITION PERIOD

A method includes determining an error condition during a bandwidth transition period of an encoded audio signal. The error condition corresponds to a second frame of the encoded audio signal, where the second frame sequentially follows a first frame in the encoded audio signal. The method also includes generating audio data corresponding to a first frequency band of the second frame based on audio data corresponding to the first frequency band of the first frame. The method further includes re-using a signal corresponding to a second frequency band of the first frame to synthesize audio data corresponding to the second frequency band of the second frame. 1. A method comprising:determining, at an electronic device during a bandwidth transition period of an encoded audio signal, an error condition corresponding to a second frame of the encoded audio signal, wherein the second frame sequentially follows a first frame in the encoded audio signal;generating audio data corresponding to a first frequency band of the second frame based on audio data corresponding to the first frequency band of the first frame; andre-using a signal corresponding to a second frequency band of the first frame to synthesize audio data corresponding to the second frequency band of the second frame.2. The method of claim 1 , wherein the bandwidth transition period corresponds to a bandwidth reduction.3. The method of claim 2 , wherein the bandwidth reduction is from:full band (FB) to super wideband (SWB);FB to wideband (WB);FB to narrowband (NB);SWB to WB;SWB to NB; orWB to NB.4. The method of claim 2 , wherein the bandwidth reduction corresponds to at least one of a reduction in encoding bitrate or a reduction in bandwidth of a signal that is encoded to generate the encoded audio signal.5. The method of claim 1 , wherein the bandwidth transition period corresponds to a bandwidth increase.6. The method of claim 1 , wherein the first frequency band includes a low-band frequency band.7. The method ...

Audio Signal Coding Method and Apparatus

The present disclosure relates to an audio signal coding method and apparatus. The method includes categorizing audio signals into high-frequency audio signals and low-frequency audio signals, coding the low-frequency audio signals using a corresponding low-frequency coding manner according to characteristics of low-frequency audio signals, and selecting a bandwidth extension mode to code the high-frequency audio signals according to the low-frequency coding manner and/or characteristics of the audio signals. 1. An audio signal coding method , comprising:categorizing audio signals into high-frequency audio signals and low-frequency audio signals;coding the low-frequency audio signals using at least one of a time domain (TD) coding manner or a frequency domain (FD) coding manner; andselecting a bandwidth extension mode to code the high-frequency audio signals according to at least one of a low-frequency coding manner or characteristics of the audio signals.2. The audio signal coding method according to claim 1 , wherein selecting the bandwidth extension mode to code the high-frequency audio signals according to the low-frequency coding manner comprises:selecting a time domain bandwidth extension (TD-BWE) mode to perform TD coding for the high-frequency audio signals when the low-frequency audio signals should be coded using the TD coding manner; andselecting a frequency domain bandwidth extension (FD-BWE) mode to perform FD coding for the high-frequency audio signals when the low-frequency audio signals should be coded using the FD coding manner.3. The audio signal coding method according to claim 1 , wherein selecting the bandwidth extension mode to code the high-frequency audio signals according to the characteristics of the audio signals comprises:selecting a time domain bandwidth extension (TD-BWE) mode to perform TD coding for the high-frequency audio signals when the audio signals are voice signals; andselecting a frequency domain bandwidth extension (FD-BWE) ...

HIGH-BAND SIGNAL GENERATION

A device for signal processing includes a memory and a processor. The memory is configured to store a parameter associated with a bandwidth-extended audio stream. The processor is configured to select a plurality of non-linear processing functions based at least in part on a value of the parameter. The processor is also configured to generate a high-band excitation signal based on the plurality of non-linear processing functions. 1. A device for signal processing comprising:a receiver configured to receive an encoded audio signal, wherein the encoded audio signal comprises a parameter;a memory configured to store the parameter associated with a bandwidth-extended audio stream; anda processor configured to:select a plurality of non-linear processing functions based at least in part on a value of the parameter, wherein the plurality of non-linear processing functions comprise a first non-linear processing function and a second non-linear processing function, wherein the first non-linear processing function is different from the second non-linear processing function;generate a first excitation signal based on the first non-linear processing function;generate a second excitation signal based on the second non-linear processing function; andgenerate a high-band excitation signal based on the first excitation signal and the second excitation signal, wherein the first excitation signal corresponds to a first high-band frequency sub-range, and wherein the second excitation signal corresponds to a second high-band frequency sub-range.2. The device of claim 1 , wherein the processor is further configured to generate a resampled signal based on a low-band excitation signal claim 1 , wherein the high-band excitation signal is based at least in part on the resampled signal.3. The device of claim 1 , wherein the processor is further configured to:generate a first filtered signal by applying a low-pass filter to the first excitation signal; andgenerate a second filtered signal by ...

METHOD FOR REDUCTION OF ALIASING INTRODUCED BY SPECTRAL ENVELOPE ADJUSTMENT IN REAL-VALUED FILTERBANKS

The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank. 1. An audio decoder for producing a full bandwidth audio signal having a lowband portion and a highband portion , the apparatus comprising:a cosine modulated, real-valued analysis filterbank that receives a time domain decoded audio signal and produces a plurality of real-valued subband signals; an aliasing detector that identifies subband signals where aliasing created by spectral envelope adjustment of an audio signal may occur based at least in part on a linear predictor applied to at least some of the plurality of real-valued subband signals;', 'an energy estimator that estimates an energy of at least some of the plurality of copied real-valued subband signals; an aliasing reducer that modifies a gain to be applied to at least some of the identified subbands signals based at least in part on the estimated energy; and', 'a real-valued synthesis filterbank that combines the plurality of real-valued subband signals with the highband portion to produce the full bandwidth audio signal, the full bandwidth audio including real-valued time domain samples,, 'a high frequency reconstructor that regenerates at least some of the highband portion by copying one or more of the plurality of real-valued subband signals up to the highband portion;'}wherein the audio deocder is implemented at least in part with one of more hardware elements, andwherein the linear predictor is a second order linear predictor.2. A method performed in an audio decoder for producing a full bandwidth audio signal having a lowband portion and a highband ...

APPARATUS, METHOD, AND NON-TRANSITORY COMPUTER-READABLE STORAGE MEDIUM FOR STORING PROGRAM FOR UTTERANCE SECTION DETECTION

A method for utterance section detection includes: executing pitch gain calculation processing that includes calculating a pitch gain indicating an intensity of periodicity of an audio signal expressing a voice of a speaker for each of frames that are obtained by dividing the audio signal and that each have a predetermined length; and executing utterance section detection processing that includes determining that an utterance section on the audio signal starts when the pitch gain becomes greater than or equal to a first threshold value after a non-utterance section on the audio signal lasts, wherein the utterance section detection processing further includes determining that the utterance section ends when the pitch gain becomes less than a second threshold value lower than the first threshold value after the utterance section lasts. 1. An apparatus for utterance section detection , the apparatus comprising:a memory; anda processor coupled to the memory and configured toexecute pitch gain calculation processing that includes calculating a pitch gain indicating an intensity of periodicity of an audio signal expressing a voice of a speaker for each of frames that are obtained by dividing the audio signal and that each have a given length, andexecute utterance section detection processing that includes determining that an utterance section on the audio signal starts when the pitch gain becomes greater than or equal to a first threshold value after a non-utterance section on the audio signal lasts, wherein the utterance section detection processing further includes determining that the utterance section ends when the pitch gain becomes less than a second threshold value lower than the first threshold value after the utterance section lasts.2. The apparatus according to claim 1 ,wherein the processor is configured toexecute signal-to-noise component ratio calculation processing that includes calculating a signal-to-noise component ratio of the audio signal for each of ...

DEEP NEURAL NETWORK BASED AUDIO PROCESSING METHOD, DEVICE AND STORAGE MEDIUM

A deep neural network based audio processing method is provided. The method includes: obtaining a deep neural network based speech extraction model; receiving an audio input object having a speech portion and a non-speech portion, wherein the audio input object includes one or more audio data frames each having a set of audio data samples sampled at a predetermined sampling interval and represented in time domain data format; obtaining a user audiogram and a set of user gain compensation coefficients associated with the user audiogram; and inputting the audio input object and the set of user gain compensation coefficients into the trained speech extraction model to obtain an audio output result represented in time domain data format outputted by the trained speech extraction model, wherein the non-speech portion of the audio input object is at least partially attenuated in or removed from the audio output result. 1. A deep neural network (DNN) based audio processing method , comprising: obtaining a mixed audio training dataset having multiple mixed audio data frames each containing mixed speech data and non-speech data, the speech data and the non-speech data both being represented in time domain data format;', 'acquiring at least one audiogram and at least one set of predetermined gain compensation coefficients associated with the at least one audiogram, wherein each audiogram corresponds to a set of predetermined gain compensation coefficients, and each set of predetermined gain compensation coefficients include multiple predetermined gain compensation coefficients corresponding to respective audio signal frequencies;', 'performing, for each of the mixed audio data frames, gain compensation on the speech data included therein with the at least one set of predetermined gain compensation coefficients to generate compensated speech data; and', 'training the DNN-based speech extraction model with the mixed audio training dataset and the compensated speech data ...

Noise filling without side information for celp-like coders

An audio decoder provides a decoded audio information on the basis of an encoded audio information including linear prediction coefficients (LPC) and includes a tilt adjuster to adjust a tilt of a noise using linear prediction coefficients of a current frame to acquire a tilt information and a noise inserter configured to add the noise to the current frame in dependence on the tilt information. Another audio decoder includes a noise level estimator to estimate a noise level for a current frame using a linear prediction coefficient of at least one previous frame to acquire a noise level information; and a noise inserter to add a noise to the current frame in dependence on the noise level information provided by the noise level estimator. Thus, side information about a background noise in the bit-stream may be omitted. Methods and computer programs serve a similar purpose.

METHOD AND APPARATUS FOR RECOVERING LOST FRAMES

A method for recovering a lost frame in a received audio signal includes: obtaining an initial high-frequency band signal of a current lost frame in the received audio signal; calculating a ratio R, wherein the ratio R is a ratio of a high frequency excitation energy of a previous frame of the current lost frame to a high frequency excitation energy of the current lost frame; obtaining a global gain of the current lost frame according to the ratio R and a global gain of the previous frame of the current lost frame; and recovering a high-frequency band signal of the current lost frame according to the initial high-frequency band signal of the current lost frame and the global gain of the current lost frame. The method can be used in an audio signal decoding process for low-loss recovery of lost frames of the audio signal. 1. A method for use by an audio signal decoder to recover lost frames in a received audio signal , comprising:obtaining an initial high-frequency band signal of a current lost frame in the received audio signal;calculating a ratio R, wherein the ratio R is a ratio of a high frequency excitation energy of a previous frame of the current lost frame to a high frequency excitation energy of the current lost frame;obtaining a global gain of the current lost frame according to the ratio R and a global gain of the previous frame of the current lost frame; andrecovering a high-frequency band signal of the current lost frame according to the initial high-frequency band signal of the current lost frame and the global gain of the current lost frame.2. The method according to claim 1 , wherein obtaining the global gain of the current lost frame according to the ratio R and the global gain of the previous frame of the current lost frame comprises: {'br': None, 'i': G′=a×R', 'a', 'G,, '+(1−)×'}, 'obtaining the global gain of the current lost frame according to the following formulawhere G′ is the global gain of the current lost frame, G is the global gain of the ...

METHOD AND ARRANGEMENT FOR CONTROLLING SMOOTHING OF STATIONARY BACKGROUND NOISE

In a method for coding of information for enhancing a background noise representation, voice activity of an input speech signal is determined. A noisiness parameter is determined for an inactive speech signal, wherein the noisiness parameter is based on a ratio of prediction gains of two Linear Predictive Coder (LPC) prediction filters with different orders. The noisiness parameter is quantized, and the quantized noisiness parameter is encoded for transmission. 1. A method of smoothing stationary background noise in a telecommunication speech session , comprising:receiving and decoding a signal representative of a speech session, said signal comprising both a speech component and a background noise component, providing a noisiness measure for said signal, said noisiness measure being indicative of the predictability of the signal, said predictability being defined in terms of an LPC prediction gain of said signal; andadaptively smoothing said background noise component based on said provided noisiness measure, wherein said smoothing operation is indirectly controlled by said noisiness measure based on a smoothing control parameter that follows a detected increase of said noisiness measure gradually, and follows a detected reduction of said noisiness measure immediately.2. The method according to claim 1 , wherein said noisiness measure is inversely dependent of the predictability3. The method according to claim 2 , wherein said noisiness measure is based on a ratio of prediction error variances associated with LPC analysis filtering with different orders.4. The method according to claim 1 , wherein said noisiness metric is adapted in response to a detected narrowband or wideband content of said input signal.5. The method according to claim 1 , wherein said noisiness providing step is performed at least once for each frame of said signal.6. The method according to claim 5 , wherein said noisiness providing step is performed for each sub-frame of each said frame of ...

Coding Method, Decoding Method, Coder, and Decoder

A coding method, a decoding method, a coder, and a decoder, where the coding method includes obtaining the pulse distribution, on a track, of the pulses to be encoded on the track, determining a distribution identifier for identifying the pulse distribution according to the pulse distribution, and generating a coding index that includes the distribution identifier. The decoding method includes receiving a coding index, obtaining a distribution identifier from the coding index, wherein the distribution identifier is configured to identify the pulse distribution, on a track, of the pulses to be encoded on the track, determining the pulse distribution, on a track, of all the pulses to be encoded on the track according to the distribution identifier, and reconstructing the pulse order on the track according to the pulse distribution.

METHOD AND APPARATUS FOR CALCULATING DOWNMIXED SIGNAL AND RESIDUAL SIGNAL

A method and an apparatus for calculating a downmixed signal and a residual signal are provided. According to the method, if a first target frame (a current frame or a previous frame of the current frame) is a switching frame, a to-be-encoded downmixed signal and a to-be-encoded residual signal of the subband corresponding to the preset frequency band in the current frame is calculated based on a switch fade-in/fade-out factor of a second target frame, an initial downmixed signal and an initial residual signal of the preset frequency band. 1. A method for calculating a downmixed signal and a residual signal , the method comprising:obtaining an initial downmixed signal and an initial residual signal of a subband corresponding to a preset frequency band in a current frame of an audio signal, which is a stereo signal;determining whether a first target frame of the audio signal is a switching frame, wherein the first target frame is the current frame or a previous frame of the current frame; andif the first target frame is a switching frame, calculating, based on a switch fade-in/fade-out factor of a second target frame, the initial downmixed signal, and the initial residual signal, a to-be-encoded downmixed signal and a to-be-encoded residual signal of the subband corresponding to the preset frequency band in the current frame,wherein the second target frame is the current frame or the previous frame of the current frame, and the switch fade-in/fade-out factor is determined based on a residual signal coding parameter of the second target frame and at least one of an inter-frame energy fluctuation parameter or an inter-frame amplitude fluctuation parameter of the second target frame, andwherein the residual signal coding parameter represents an energy relationship between a first downmixed signal and a first residual signal of the second target frame, and the inter-frame energy fluctuation parameter or the inter-frame amplitude fluctuation parameter represents an energy ...

SPEECH PROCESSING METHOD AND DEVICE THEREOF

The disclosure provides a speech processing method and a device thereof. The method includes: acquiring a speech sampling signal frame in a mixed-excitation linear prediction (MELP) speech coding system and estimating signal quality of the speech sampling signal frame; determining, based on the signal quality, a specific linear prediction coding (LPC) order used by an LPC circuit; controlling the LPC circuit to convert the speech sampling signal frame into a line spectrum pair parameter based on the specific LPC order; replacing a speech signal spectrum of the speech sampling signal frame with the line spectrum pair parameter to generate a predicted speech signal; and performing a speech coding operation and a signal synthesizing operation of the MELP speech coding system based on the predicted speech signal. 1. A speech processing method , comprising:acquiring a speech sampling signal frame in a mixed-excitation linear prediction speech coding system and estimating signal quality of the speech sampling signal frame, wherein the mixed-excitation linear prediction speech coding system comprises a linear prediction coding circuit;determining, based on the signal quality, a specific linear prediction coding order used by the linear prediction coding circuit;controlling the linear prediction coding circuit to convert the speech sampling signal frame into a line spectrum pair parameter based on the specific linear prediction coding order;replacing a speech signal spectrum of the speech sampling signal frame with the line spectrum pair parameter to generate a predicted speech signal; andperforming a speech coding operation and a signal synthesizing operation of the mixed-excitation linear prediction speech coding system based on the predicted speech signal.2. The method according to claim 1 , wherein the signal quality is represented as a signal to interference plus noise ratio of the speech sampling signal frame.3. The method according to claim 1 , wherein the step of ...

Method for reduction of aliasing introduced by spectral envelope adjustment in real-valued filterbanks

The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

OPTIMIZED SCALE FACTOR FOR FREQUENCY BAND EXTENSION IN AN AUDIO FREQUENCY SIGNAL DECODER

A method and device are provided for determining an optimized scale factor to be applied to an excitation signal or a filter during a process for frequency band extension of an audio frequency signal. The band extension process includes decoding or extracting, in a first frequency band, an excitation signal and parameters of the first frequency band including coefficients of a linear prediction filter, generating an excitation signal extending over at least one second frequency band, filtering using a linear prediction filter for the second frequency band. The determination method includes determining an additional linear prediction filter, of a lower order than that of the linear prediction filter of the first frequency band, the coefficients of the additional filter being obtained from the parameters decoded or extracted from the first frequency and calculating the optimized scale factor as a function of at least the coefficients of the additional filter. 1. A method for determining an optimized scale factor to be applied to an excitation signal or to a filter in a method of extending a frequency band of an audio frequency signal ,the method comprising steps of:computing of a frequency response, R, of a linear prediction filter of a frequency band,{'sub': 'smoothed', 'smoothing of the value of R, so as to obtain R, the smoothing method being selected, from a group of smoothing methods including at least two smoothing methods, in function of a set of parameters comprising a plurality of parameters including the value of spectral slope, tilt, wherein the set of smoothing methods comprises a smoothing method being adaptive over time.'}2. The method of claim 1 , wherein the smoothing is stronger for smaller values of R.3. The method of claim 1 , wherein the adaptive smoothing is of the form:{'br': None, 'i': R', 'R', '+α·R', 'R, 'sub': smoothed', 'precomputed', 'prev', 'precomputed, '=(1−α), where α=1−̂2.'}{'sub': prev', 'smoothed', 'precomputed, 'where Rcorresponds ...

Signal Decoding Method and Device

Embodiments of the present invention provide a signal decoding method and device. The method includes decoding a bit stream of a voice signal or an audio signal to acquire a decoded signal; predicting an excitation signal of an extension band according to the decoded signal, where the extension band is adjacent to a band of the decoded signal, and the band of the decoded signal is lower than the extension band; selecting a first band and a second band from the decoded signal, and predicting a spectral envelope of the extension band according to a spectral coefficient of the first band and a spectral coefficient of the second band; and determining a frequency-domain signal of the extension band according to the spectral envelope of the extension band and the excitation signal of the extension band. 1. A signal decoding method , comprising:decoding a bit stream of a voice signal or an audio signal to acquire a decoded signal;predicting an excitation signal of an extension band according to the decoded signal, wherein the extension band is adjacent to a band of the decoded signal, and wherein the band of the decoded signal is lower than the extension band;selecting a first band and a second band from the decoded signal;predicting a spectral envelope of the extension band according to a spectral coefficient of the first band and a spectral coefficient of the second band, wherein a distance from a highest frequency bin of the first band to a lowest frequency bin of the extension band is less than or equal to a first value, and a distance from a highest frequency bin of the second band to a lowest frequency bin of the first band is less than or equal to a second value; anddetermining a frequency-domain signal of the extension band according to the spectral envelope of the extension band and the excitation signal of the extension band.2. The method according to claim 1 , wherein selecting the first band and the second band comprises selecting the first band and the second ...

HIGH-BAND SIGNAL CODING USING MULTIPLE SUB-BANDS

A method includes receiving, at a first device, a bit-stream from a second device. The method also includes generating, at a decoder of the first device, a low-band excitation signal from the bit-stream. The method also includes generating a first baseband signal at a high-band excitation generator of the decoder. Generating the first baseband signal includes performing a spectral flip operation on a nonlinearly transformed version of the low-band excitation signal, and the first baseband signal corresponds to a first sub-band of a high-band portion of an audio signal received at the second device. The method also includes generating a second baseband signal corresponding to a second sub-band of the high-band portion of the audio signal. The method also includes outputting at least a partially reconstructed version of the audio signal based at least in part on the first baseband signal and the second baseband signal. 1. A method comprising:receiving, at a first device, a bit-stream from a second device;generating, at a decoder of the first device, a low-band excitation signal from the bit-stream;generating a first baseband signal at a high-band excitation generator of the decoder, wherein generating the first baseband signal includes performing a spectral flip operation on a nonlinearly transformed version of the low-band excitation signal, the first baseband signal corresponding to a first sub-band of a high-band portion of an audio signal received at the second device;generating a second baseband signal corresponding to a second sub-band of the high-band portion of the audio signal, wherein the first sub-band is distinct from the second sub-band; andoutputting at least a partially reconstructed version of the audio signal based at least in part on the first baseband signal and the second baseband signal.2. The method of claim 1 , wherein the second baseband signal is generated based on the first baseband signal.3. The method of claim 2 , wherein generating the ...

HIGH-BAND SIGNAL GENERATION

A device for signal processing includes a memory and a processor. The memory is configured to store a parameter associated with a bandwidth-extended audio stream. The processor is configured to select a plurality of non-linear processing functions based at least in part on a value of the parameter. The processor is also configured to generate a high-band excitation signal based on the plurality of non-linear processing functions.

Audio signal encoding and decoding

An audio codec suitable for robust wireless transmission of high quality audio with low latency, still at a moderate bit rate. The encoding and decoding methods are based on ADPCM and in addition to the encoded output bits APM, additional data QB are included in output data blocks, namely data QB representing an internal value of the adaptive quantization ADQ of the ADPCM encoding algorithm, especially a scaling factor encoded and truncated to such as 8 bits. Further, output data blocks preferably include data CFB representing an internal value of the predictor PR of the ADPCM encoding algorithm, especially data CFB representing coefficients of a lattice prediction FIR filter which, truncated to such as 8 bits, can be sequentially included in output data blocks. These additional data QB, CFB regarding internal values of the ADPCM encoding algorithm can be utilized at the encoder side to increase robustness against loss of data blocks in wireless transmission. Especially, the decoding algorithm may comprise comparing its current internal ADPCM decoding values corresponding to the received internal values QB, CFB from the encoder, and in case there is a difference, the decoder can adapt or overwrite its internal values to the ones received QB, CFB. This helps to ensure fast recovery after lost data blocks, thereby ensuring robustness against artefacts in the reconstructed signal, e.g. clicks in case of audio.

APPARATUS AND METHOD FOR IMPROVED SIGNAL FADE OUT IN DIFFERENT DOMAINS DURING ERROR CONCEALMENT

An apparatus for decoding an audio signal is provided, having a receiving interface, configured to receive a first frame having a first audio signal portion of the audio signal, and configured to receive a second frame having a second audio signal portion of the audio signal; a noise level tracing unit, wherein the noise level tracing unit is configured to determine noise level information depending on at least one of the first audio signal portion and the second audio signal portion; a first reconstruction unit for reconstructing, in a first reconstruction domain, a third audio signal portion of the audio signal depending on the noise level information; a transform unit for transforming the noise level information to a second reconstruction domain; and a second reconstruction unit for reconstructing, in the second reconstruction domain, a fourth audio signal portion of the audio signal depending on the noise level information. 1. An apparatus for decoding an audio signal , comprising:a receiving interface, wherein the receiving interface is configured to receive a first frame comprising a first audio signal portion of the audio signal, and wherein the receiving interface is configured to receive a second frame comprising a second audio signal portion of the audio signal,a noise level tracing unit, wherein the noise level tracing unit is configured to determine noise level information depending on at least one of the first audio signal portion and the second audio signal portion, wherein the noise level information is represented in a tracing domain,a first reconstruction unit for reconstructing, in a first reconstruction domain, a third audio signal portion of the audio signal depending on the noise level information, if a third frame of the plurality of frames is not received by the receiving interface or if said third frame is received by the receiving interface but is corrupted, wherein the first reconstruction domain is different from or equal to the tracing ...

CONCEPT FOR ENCODING AN AUDIO SIGNAL AND DECODING AN AUDIO SIGNAL USING SPEECH RELATED SPECTRAL SHAPING INFORMATION

According to an aspect of the present invention an encoder for encoding an audio signal has an analyzer configured for deriving prediction coefficients and a residual signal from a frame of the audio signal. The encoder has a formant information calculator configured for calculating a speech related spectral shaping information from the prediction coefficients, a gain parameter calculator configured for calculating a gain parameter from an unvoiced residual signal and the spectral shaping information and a bitstream former configured for forming an output signal based on an information related to a voiced signal frame, the gain parameter or a quantized gain parameter and the prediction coefficients. 1. An encoder for encoding an audio signal , the encoder comprisingan analyzer configured for deriving prediction coefficients and a residual signal from a frame of the audio signal;a formant information calculator configured for calculating a speech related spectral shaping information from the prediction coefficients;a gain parameter calculator configured for calculating a gain parameter from an unvoiced residual signal and the spectral shaping information; anda bitstream former configured for forming an output signal based on an information related to a voiced signal frame, the gain parameter or a quantized gain parameter and the prediction coefficients.2. The encoder according to claim 1 , further comprising a decider configured for determining if the residual signal was determined from an unvoiced signal audio frame.3. The encoder according to claim 1 , wherein the gain parameter calculator comprises:a noise generator configured for generating an encoding noise-like signal;a shaper configured for amplifying and shaping a spectrum of the encoding noise-like signal using the speech related spectral shaping information and the gain parameter as temporary gain parameter to acquire an amplified shaped encoding noise-like signal;a comparer configured for comparing the ...

APPARATUS AND METHOD FOR IMPROVED SIGNAL FADE OUT FOR SWITCHED AUDIO CODING SYSTEMS DURING ERROR CONCEALMENT

An apparatus for decoding an audio signal includes a receiving interface, wherein the receiving interface is configured to receive a first frame and a second frame. Moreover, the apparatus includes a noise level tracing unit for determining noise level information being represented in a tracing domain. Furthermore, the apparatus includes a first reconstruction unit for reconstructing a third audio signal portion of the audio signal depending on the noise level information and a second reconstruction unit for reconstructing a fourth audio signal portion depending on noise level information being represented in the second reconstruction domain. 1. An apparatus for decoding an audio signal , comprising:a receiving interface for receiving a plurality of frames, wherein the receiving interface is configured to receive a first frame of the plurality of frames, said first frame comprising a first audio signal portion of the audio signal, said first audio signal portion being represented in a first domain, and wherein the receiving interface is configured to receive a second frame of the plurality of frames, said second frame comprising a second audio signal portion of the audio signal,a transform unit for transforming the second audio signal portion or a value or signal derived from the second audio signal portion from a second domain to a tracing domain to acquire a second signal portion information, wherein the second domain is different from the first domain, wherein the tracing domain is different from the second domain, and wherein the tracing domain is equal to or different from the first domain,a noise level tracing unit, wherein the noise level tracing unit is configured to receive a first signal portion information being represented in the tracing domain, wherein the first signal portion information depends on the first audio signal portion, wherein the noise level tracing unit is configured to receive the second signal portion being represented in the tracing ...

APPARATUS AND METHOD REALIZING IMPROVED CONCEPTS FOR TCX LTP

An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal is provided. The apparatus includes a receiving interface, a delay buffer and a sample processor for processing the selected audio signal samples to obtain reconstructed audio signal samples of the reconstructed audio signal. The sample selector is configured to select, if a current frame is received by the receiving interface and if the current frame being received by the receiving interface is not corrupted, the plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer depending on a pitch lag information being included by the current frame. 1. An apparatus for decoding an encoded audio signal to acquire a reconstructed audio signal , wherein the apparatus is configured to receive a plurality of frames , and wherein the apparatus comprises:an inverse modified discrete cosine transform module for decoding the plurality of frames by conducting an inverse modified discrete cosine transform to acquire audio signal samples of the decoded audio signal, and a delay buffer for storing the audio signal samples of the decoded audio signal,', 'a sample selector for selecting a plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer, and', 'a sample processor for processing the selected audio signal samples to acquire reconstructed audio signal samples of the reconstructed audio signal,, 'a long-term prediction unit for conducting long-term prediction, comprisingwherein the sample selector is configured to select, if a current frame is received by the apparatus and if the current frame being received by the apparatus is not corrupted, the plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer depending on a pitch lag information being comprised by the current frame, andwherein the sample selector is configured to select, if the current frame ...

METHOD AND APPARATUS FOR PROCESSING LOST FRAME

Embodiments of the present application provide a method and an apparatus for recovering a lost frame in a received audio signal. The method for recovering a lost frame includes: determining an initial high-frequency band signal of a current lost frame; determining a gain of the current lost frame; determining gain adjustment information of the current lost frame; adjusting the gain of the current lost frame according to the gain adjustment information, to obtain an adjusted gain of the current lost frame; and adjusting the initial high-band signal according to the adjusted gain, to obtain a high-frequency band signal of the current lost frame. The method and the apparatus for recovering a lost frame provided in the embodiments of the present application can be used in an audio signal decoding process for low-loss recovery of a lost frame of the audio signal, resulting in improved performance of an audio signal decoder. 1. A method for recovering lost frames in a received audio signal , performed by a signal decoding apparatus in an audio signal receiver , wherein the method comprises:determining an initial high-frequency band signal of a current lost frame in a received audio signal;determining a gain of the current lost frame according to the gain of a previous frame of the current lost frame;obtaining adjustment information of the current lost frame for adjusting the gain of the current lost frame;obtaining an adjusted gain of the current lost frame by adjusting the gain of the current lost frame according to the adjustment information; andobtaining a high-frequency band signal of the current lost frame by adjusting the initial high-frequency band signal of the lost frame according to the adjusted gain; class information of the current lost frame,', 'low-frequency band spectral tilt of the current lost frame,', 'low-frequency band signal energy of the current lost frame, and', 'a quantity of consecutive frames that are lost, including the current lost frame., ' ...

APPARATUS AND METHOD FOR SELECTING ONE OF A FIRST ENCODING ALGORITHM AND A SECOND ENCODING ALGORITHM

An apparatus for selecting one of a first encoding algorithm having a first characteristic and a second encoding algorithm having a second characteristic for encoding a portion of an audio signal to obtain an encoded version of the portion of the audio signal has a first estimator for estimating a first quality measure for the portion of the audio signal, which is associated with the first encoding algorithm, without actually encoding and decoding the portion of the audio signal using the first encoding algorithm. A second estimator is provided for estimating a second quality measure for the portion of the audio signal, which is associated with the second encoding algorithm, without actually encoding and decoding the portion of the audio signal using the second encoding algorithm. The apparatus has a controller for selecting the first or second encoding algorithms based on a comparison between the first and second quality measures. 1. An apparatus for selecting one of a first encoding algorithm comprising a first characteristic and a second encoding algorithm comprising a second characteristic for encoding a portion of an audio signal to acquire an encoded version of the portion of the audio signal , comprising:a first estimator for estimating a first quality measure for the portion of the audio signal, the first quality measure being associated with the first encoding algorithm, without actually encoding and decoding the portion of the audio signal using the first encoding algorithm;a second estimator for estimating a second quality measure for the portion of the audio signal, the second quality measure being associated with the second encoding algorithm, without actually encoding and decoding the portion of the audio signal using the second encoding algorithm; anda controller for selecting the first encoding algorithm or the second encoding algorithm based on a comparison between the first quality measure and the second quality measure,wherein the first and second ...

Apparatus and method for concealing frame erasure and voice decoding apparatus and method using the same

An apparatus and method for concealing frame erasure and a voice decoding apparatus and method using the same. The frame erasure concealment apparatus includes: a parameter extraction unit determining whether there is an erased frame in a voice packet, and extracting an excitement signal parameter and a line spectrum pair parameter of a previous good frame; and an erasure frame concealment unit, if there is an erased frame, restoring the excitement signal and line spectrum pair parameter of the erased frame by using a regression analysis from the excitement signal and line spectrum pair parameter of the previous good frame. According to the method and apparatus, by predicting and restoring the parameter of the erased frame through the regression analysis, the quality of the restored voice signal can be enhanced and the algorithm can be simplified.

METHOD FOR REDUCTION OF ALIASING INTRODUCED BY SPECTRAL ENVELOPE ADJUSTMENT IN REAL-VALUED FILTERBANKS

The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank. 1. Apparatus for performing gain adjustment on a plurality of audio sub band signals generated by filtering an audio signal using a filter bank , the filter bank having sub band filters , adjacent sub band filters of the filterbank having transition bands overlapping in an overlapping range , comprising:an analyzer for analysing the plurality of audio sub band signals generated by filtering the signal using the filter bank to determine, whether a subband signal of a sub band filter and a sub band signal of an adjacent sub band filter have aliasing generating signal components in the overlapping range between the sub band filter and the adjacent sub band filter to obtain grouped audio sub band signals; to determine a first energy measure indicating a signal energy of the audio sub band signal and a second energy measure indicating a signal energy of the adjacent audio sub band signal,', 'to determine an indication of a reference energy for the grouped adjacent audio sub band signals as a linear combination of a first reference energy value for the audio sub band signal and a second reference energy value for the adjacent audio sub band signal, and', 'to determine an energy estimate for an energy in the grouped adjacent audio sub band signals as a linear combination of the first energy measure for the audio sub band signal and the second energy measure for the adjacent audio sub band signal, and', 'to calculate the first gain adjustment value and the second gain adjustment value for the grouped adjacent audio sub band signals ...

Deep neural network-based method and apparatus for combining noise and echo removal

Disclosed is a deep neural network-based method and apparatus for combining noise and echo removal. The deep neural network-based method for combining noise and echo removal according to one embodiment of the present invention may comprise the steps of extracting a feature vector from an audio signal that includes noise and echo; and acquiring a final audio signal from which both noise and echo have been removed, by using a combined nose and echo removal gain estimated by means of the feature vector and deep neural network DNN.

Apparatus and method for improved concealment of the adaptive codebook in a celp-like concealment employing improved pitch lag estimation

An apparatus for determining an estimated pitch lag is provided. The apparatus includes an input interface for receiving a plurality of original pitch lag values, and a pitch lag estimator for estimating the estimated pitch lag. The pitch lag estimator is configured to estimate the estimated pitch lag depending on a plurality of original pitch lag values and depending on a plurality of information values, wherein for each original pitch lag value of the plurality of original pitch lag values, an information value of the plurality of information values is assigned to the original pitch lag value.

AUDIO DECODER AND METHOD FOR PROVIDING A DECODED AUDIO INFORMATION USING AN ERROR CONCEALMENT BASED ON A TIME DOMAIN EXCITATION SIGNAL

An audio decoder and method for providing a decoded audio information on the basis of an encoded audio information are disclosed. In one example, the audio decoder includes an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame following an audio frame encoded in a frequency domain representation using a time domain excitation signal. 1. An audio decoder for providing decoded audio information on the basis of an encoded audio information , the audio decoder comprising:an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame following an audio frame encoded in a frequency domain representation using a time domain excitation signal;wherein the frequency domain representation comprises an encoded representation of a plurality of spectral values and an encoded representation of a plurality of scale factors for scaling the spectral values, and wherein the audio decoder is configured to provide a plurality of decoded scale factors for scaling spectral values on the basis of a plurality of encoded scale factors, orwherein the audio decoder is configured to derive a plurality of scale factors for scaling the spectral values from an encoded representation of LPC parameters; andwherein the error concealment is configured to acquire the time domain excitation signal on the basis of the audio frame encoded in the frequency domain representation preceding a lost audio frame.2. A method for providing a decoded audio information on the basis of an encoded audio information , the method comprising:providing an error concealment audio information for concealing a loss of an audio frame following an audio frame encoded in a frequency domain representation using a time domain excitation signal;wherein the frequency domain representation comprises an encoded representation of a plurality of spectral values and an encoded representation of a plurality of ...

APPARATUS AND METHOD REALIZING IMPROVED CONCEPTS FOR TCX LTP

An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal is provided. The apparatus includes a receiving interface, a delay buffer and a sample processor for processing the selected audio signal samples to obtain reconstructed audio signal samples of the reconstructed audio signal. The sample selector is configured to select, if a current frame is received by the receiving interface and if the current frame being received by the receiving interface is not corrupted, the plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer depending on a pitch lag information being included by the current frame. 1. An apparatus for decoding an encoded audio signal to acquire a reconstructed audio signal , wherein the apparatus comprises:a receiving interface for receiving a plurality of frames,a delay buffer for storing audio signal samples of the decoded audio signal,a sample selector for selecting a plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer, anda sample processor for processing the selected audio signal samples to acquire reconstructed audio signal samples of the reconstructed audio signal,wherein the sample selector is configured to select, if a current frame is received by the receiving interface and if the current frame being received by the receiving interface is not corrupted, the plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer depending on a pitch lag information being comprised by the current frame, andwherein the sample selector is configured to select, if the current frame is not received by the receiving interface or if the current frame being received by the receiving interface is corrupted, the plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer depending on a pitch lag information being comprised by another frame ...

RESIDUAL CODING METHOD OF LINEAR PREDICTION CODING COEFFICIENT BASED ON COLLABORATIVE QUANTIZATION, AND COMPUTING DEVICE FOR PERFORMING THE METHOD

Disclosed are a method for coding a residual signal of LPC coefficients based on collaborative quantization and a computing device for performing the method. The residual signal coding method includes: generating encoded LPC coefficients and LPC residual signals by performing LPC analysis and quantization on an input speech; Determining a predicted LPC residual signal by applying the LPC residual signal to cross module residual learning; Performing LPC synthesis using the coded LPC coefficients and the predicted LPC residual signal; It may include the step of determining an output speech that is a synthesized output according to a result of performing the LPC synthesis. 1. A residual signal coding method of LPC (Linear Prediction Coding) coefficients performed by a computing device , the residual signal coding method comprising:generating coded LPC coefficients and LPC residual signals by performing, by a computing device, LPC analysis and quantization on an input speech;determining a predicted LPC residual signal by applying the LPC residual signal to cross module residual learning;performing LPC synthesis using the coded LPC coefficients and the predicted LPC residual signal;determining an output speech that is a synthesized output according to the result of performing the LPC synthesis.2. The residual signal coding method of claim 1 , wherein the cross module residual learning including:applying a high-pass filter to the input speech;applying a pre-emphasis filter to a result applied by the high pass filter;determining the LPC coefficient from the result of applying the pre-emphasis filter;generating a soft assignment matrix of the coded LPC coefficients and softmax by quantizing the LPC coefficients; anddetermining an LPC residual signal based on a result of applying the pre-emphasis filter and a result of quantizing the LPC coefficients.3. The residual signal coding method of claim 1 , wherein the determining the LPC coefficient claim 1 , comprising:performing ...

Method for Predicting Bandwidth Extension Frequency Band Signal, and Decoding Device

A method for predicting a bandwidth extension frequency band signal includes demultiplexing a received bitstream to obtain a frequency domain signal; determining whether a highest frequency bin, to which a bit is allocated, of the frequency domain signal is less than a preset start frequency bin of a bandwidth extension frequency band; predicting an excitation signal of the bandwidth extension frequency band according to the determination; and predicting the bandwidth extension frequency band signal according to the predicted excitation signal of the bandwidth extension frequency band and a frequency envelope of the bandwidth extension frequency band. 1. A method for predicting a bandwidth extension frequency band signal of an audio signal , comprising:obtaining, by a decoder, a decoded signal of a low frequency part of a current frame of the audio signal based on a received bitstream, wherein the audio signal comprises a plurality of frames;determining, by the decoder, whether a highest frequency bin of the decoded signal is less than a preset start frequency bin for bandwidth extension;when the highest frequency bin of the decoded signal is less than the preset start frequency bin for bandwidth extension, predicting, by the decoder, an excitation signal of a bandwidth extension signal of a high frequency part of the current frame based on an excitation signal within a predetermined frequency range of the decoded signal and the preset start frequency bin for bandwidth extension;reconstructing, by the decoder, the bandwidth extension signal of a high frequency part of the current frame based on the predicted excitation signal; andobtaining, by the decoder, a frequency domain signal of the current frame based on the decoded signal of the low frequency part of the current frame and the reconstructed bandwidth extension signal of the high frequency part of the current frame.2. The method according to claim 1 , wherein the highest frequency bin of the decoded signal is ...

Audio coding device, audio coding method, audio coding program, audio decoding device, audio decoding method, and audio decoding program

An audio signal transmission device for encoding an audio signal includes an audio encoding unit that encodes an audio signal and a side information encoding unit that calculates and encodes side information from a look-ahead signal. An audio signal receiving device for decoding an audio code and outputting an audio signal includes: an audio code buffer that detects packet loss based on a received state of an audio packet, an audio parameter decoding unit that decodes an audio code when an audio packet is correctly received, a side information decoding unit that decodes a side information code when an audio packet is correctly received, a side information accumulation unit that accumulates side information obtained by decoding a side information code, an audio parameter missing processing unit that outputs an audio parameter upon detection of audio packet loss, and an audio synthesis unit that synthesizes decoded audio from the audio parameter.

IMPROVED FREQUENCY BAND EXTENSION IN AN AUDIO SIGNAL DECODER

The invention relates to a method for extending the frequency band of an audio signal during a decoding or improvement process comprising a step of decoding or extracting, in a first so-called low frequency band, an excitation signal and coefficients of a linear prediction filter. The method comprises the following steps: —obtaining a signal (U(k), E403)) extended in at least a second frequency band higher than the first frequency band from an oversampled excitation signal extended in at least a second frequency band (UHB1(k), E401); —scaling (E406) the extended signal by means of a gain defined by subframe on the basis of an energy ratio of a frame and of a subframe; —filtering (E404) said scaled extended signal with a linear prediction filter of which the coefficients are derived from the coefficients of the low frequency band filter. The invention also relates to a frequency band extension device implementing the described method and a decoder comprising such a device. 1. A method for extending the frequency band of an audio frequency signal in a decoding or enhancement process comprising a step of decoding or of extraction , in a first frequency band called low band , of an excitation signal and of the coefficients of a linear prediction filter , the method being characterized in that it comprises the following steps:{'sub': HB2', 'HB1, 'b': 403', '401, 'obtaining of an extended signal (U(k), E)) in at least one second frequency band higher than the first frequency band from the excitation signal oversampled and extended in the at least one second frequency band (U(k), E);'}{'b': '406', 'scaling (E) of the extended signal by a gain defined per sub-frame as a function of a ratio of energy of a frame and of a sub-frame;'}{'b': '404', 'filtering (E) of said scaled extended signal by a linear prediction filter whose coefficients are derived from the coefficients of the low-band filter.'}2405. The method as claimed in claim 1 , characterized in that it further ...

AUDIO ENCODER AND DECODER USING A FREQUENCY DOMAIN PROCESSOR , A TIME DOMAIN PROCESSOR, AND A CROSS PROCESSING FOR CONTINUOUS INITIALIZATION

An audio encoder for encoding an audio signal, includes: a first encoding processor for encoding a first audio signal portion in a frequency domain, wherein the first encoding processor includes: a time frequency converter for converting the first audio signal portion into a frequency domain representation having spectral lines up to a maximum frequency of the first audio signal portion; a spectral encoder for encoding the frequency domain representation; a second encoding processor for encoding a second different audio signal portion in the time domain; a cross-processor for calculating, from the encoded spectral representation of the first audio signal portion, initialization data of the second encoding processor, so that the second encoding processing is initialized to encode the second audio signal portion immediately following the first audio signal portion in time in the audio signal. 1. An audio encoder for encoding an audio signal , comprising: a time frequency converter for converting the first audio signal portion into a frequency domain representation comprising spectral lines up to a maximum frequency of the first audio signal portion;', 'a spectral encoder for encoding the frequency domain representation;, 'a first encoding processor for encoding a first audio signal portion in a frequency domain, wherein the first encoding processor comprisesa second encoding processor for encoding a second different audio signal portion in the time domain,wherein the second encoding processor comprises an associated second sampling rate,wherein the first encoding processor has associated therewith a first sampling rate being different from the second sampling rate; a selector for selecting a portion of a spectrum input into the frequency time converter in accordance with a ratio of the first sampling rate and the second sampling rate,', 'a transform processor comprising a transform length being different from a transform length of the time-frequency converter; and', ' ...

Methods and devices for generation and processing of modified audio bitstreams

Described herein is a method for generating a modified bitstream on a source device, wherein the method includes the steps of: a) receiving, by a receiver, a bitstream including coded media data; b) generating, by an embedder, payload of additional media data and embedding the payload in the bitstream for obtaining, as an output from the embedder, a modified bitstream including the coded media data and the payload of the additional media data; and c) outputting the modified bitstream to a sink device. Described is further a method for processing said modified bitstream on a sink device. Described are moreover a respective source device and sink device as well as a system of a source device and a sink device and respective computer program products.

Electroacoustic conversion chain with selectively powered coil

This electroacoustic conversion chain comprising at least one loudspeaker, this loudspeaker comprising means for generating a magnetic field in a magnetic circuit having an air gap and a membrane secured to turns of a conducting material which may move in this air gap, is characterized in that it comprises at least one control module comprising at least one input for conveying a signal to be broadcast and at least one output connected to a turn, the control module being able to apply to said or each output an excitation signal depending on the position of at least one turn relatively to the air gap.

Method and Arrangement for Scalable Low-Complexity Coding/Decoding

In a quantization method for quantizing a received excitation signal in a communication system performing the steps of re-shuffling S the elements of the received excitation signal to provide a re-shuffled excitation signal; coding S the re-shuffled excitation signal with a variable bit-rate algorithm to provide a coded excitation signal; and reassigning S codewords of the coded excitation signal if a number of used bits exceeds a predetermined fixed bit rate requirement to provide a quantized excitation signal. 125-. (canceled)26. A quantization method for quantizing a received excitation signal in a communication system , comprising the steps of:re-shuffling the elements of said received excitation signal to provide a re-shuffled excitation signal;coding (the re-shuffled excitation signal with a variable bit-rate algorithm to provide a coded excitation signal;reassigning codewords of said coded excitation signal if a number of used bits exceeds a predetermined fixed bit rate requirement to provide a quantized excitation signal.27. The quantization method according to claim 26 , comprising performing said coding step on the elements of the received excitation signal and prior to performing said re-shuffling step on the coded excitation signal.28. The quantization method according to claim 26 , wherein said coding step comprises both SQ coding and entropy coding the re-shuffled excitation signal.29. The quantization method according to claim 28 , further comprising the step of inversely re-shuffling the coded excitation signal after said step of codeword reassignment.30. A quantizer unit for quantizing a received excitation signal a communication system claim 28 , comprising:a re-shuffling unit configured for re-shuffling the elements of said received excitation signal to provide a re-shuffled excitation signal;a coding unit configured for coding the re-shuffled excitation signal with a variable bit-rate algorithm to provide a coded excitation signal;a reassigning ...

AUDIO SIGNAL CODING METHOD AND APPARATUS

The present invention relates to an audio signal coding method and apparatus. The method includes: categorizing audio signals into high-frequency audio signals and low-frequency audio signals; coding the low-frequency audio signals by using a corresponding low-frequency coding manner according to characteristics of low-frequency audio signals; and selecting a bandwidth extension mode to code the high-frequency audio signals according to the low-frequency coding manner and/or characteristics of the audio signals. 1. An audio signal coding method , comprising:categorizing audio signals into high-frequency audio signals and low-frequency audio signals;coding the low-frequency audio signals by using a time domain coding manner or a frequency domain coding manner; andselecting a bandwidth extension mode to code the high-frequency audio signals according to a low-frequency coding manner or characteristics of the audio signals.2. The audio signal coding method according to claim 1 , wherein the selecting the bandwidth extension mode to code the high-frequency audio signals according to the low-frequency coding manner comprises determining that the low-frequency audio signals should be coded by using the time domain coding manner claim 1 , selecting a time domain bandwidth extension mode to perform time domain coding for the high-frequency audio signals.3. The audio signal coding method according to claim 1 , wherein the selecting the bandwidth extension mode to code the high-frequency audio signals according to the low-frequency coding manner comprises determining that the frequency domain bandwidth extension mode to perform frequency domain coding for the high-frequency audio signals.3. The audio signal coding method according to claim 1 , wherein the selecting the bandwidth extension mode to code the high-frequency audio signals according to the characteristics of the audio signals specifically is: if the audio signals are voice signals claim 1 , selecting a time domain ...

Training Apparatus, Speech Synthesis System, and Speech Synthesis Method

A training apparatus includes an autoregressive model configured to estimate a current signal from a past signal sequence and a current context label, a vocal tract feature analyzer configured to analyze an input speech signal to determine a vocal tract filter coefficient representing a vocal tract feature, a residual signal generator configured to output a residual signal, a quantization unit configured to quantize the residual signal output from the residual signal generator to generate a quantized residual signal, and a training controller configured to provide as a condition, a context label of an already known input text for the input speech signal corresponding to the already known input text to the autoregressive model and to train the autoregressive model by bringing a past sequence of the quantized residual signals for the input speech signal and the current context label into correspondence with a current signal of the quantized residual signal.

SIGNALING PROCESSOR AND CONTROL METHOD THEREOF

A signaling processor is provided. The signaling processor includes a frequency domain processing module configured to generate a cut-off frequency of an input signal and to generate level information for adjusting a level of a high frequency recovery signal and a time domain processing module configured to receive the cut-off frequency and the level information from the frequency domain processing module, to generate a signal having a frequency greater than or equal to the cut-off frequency using part of a signal of a frequency lower than the cut-off frequency in the input signal, to generate the high frequency recovery signal by adjusting a level of the generated signal using the level information, and to synthesize the high frequency recovery signal with the input signal. 1. A signaling processor , comprising:a frequency domain processing module comprising processing circuitry configured to generate a cut-off frequency of an input signal and to generate level information for adjusting a level of a high frequency recovery signal; anda time domain processing module comprising processing circuitry configured to receive the cut-off frequency and the level information from the frequency domain processing module, to generate a signal having a frequency greater than or equal to the cut-off frequency using part of a signal having a frequency lower than the cut-off frequency in the input signal, to generate the high frequency recovery signal by adjusting a level of the generated signal using the level information, and to synthesize the high frequency recovery signal with the input signal.2. The signaling processor of claim 1 , wherein the input signal comprises additional information about a format of the input signal claim 1 , andwherein the frequency domain processing module is configured to:generate the cut-off frequency in a frequency band corresponding to the additional information.3. The signaling processor of claim 2 , wherein the frequency domain processing module ...

SPATIALLY BIASED SOUND PICKUP FOR BINAURAL VIDEO RECORDING

A method for producing a target directivity function that includes a set of spatially biased HRTFs. A set of left ear and right ear head related transfer functions (HRTFs) are selected. The left ear and right ear head HRTFs are multiplied with an on-camera emphasis function (OCE), to produce the spatially biased HRTFs. The OCE may be designed to shape the sound profile of the HRTFs to provide emphasis in a desired location or direction that is a function of the specific orientation of the device as it is being used to make a video recording. Other aspects are also described and claimed. 1. A method for producing a spatially biased sound pickup beamforming function , the method comprising:generating a target directivity function that includes a set of spatially biased head related transfer functions;generating a left ear set of beamforming coefficients and a right ear set of beamforming coefficients by determining a fit for the target directivity function based on a device steering matrix; andoutputting the left ear set of beamforming coefficients and the right ear set of beamforming coefficients.2. The method of claim 1 , wherein the device steering matrix includes a plurality of transfer functions of a plurality of microphones claim 1 , wherein each of the transfer functions describes a response by a respective one of the microphones to a single sound source direction.3. The method of claim 2 , wherein the fit for the target directivity coefficients is determined by utilizing a least squares method claim 2 , wherein the least squares method comprises inputting the target directivity function and the device steering matrix into a least squares beamformer design algorithm.4. The method of claim 3 , wherein the least squares beamformer design algorithm includes a determined white-noise gain constraint while determining a fit for the target directivity function based on a device steering matrix for a first ear.5. The method of claim 4 , wherein the least squares method ...

Coding method, decoding method, coder, and decoder

A coding method, a decoding method, a coder, and a decoder are disclosed herein. A coding method includes: obtaining the pulse distribution, on a track, of the pulses to be encoded on the track; determining a distribution identifier for identifying the pulse distribution according to the pulse distribution; and generating a coding index that includes the distribution identifier. A decoding method includes: receiving a coding index; obtaining a distribution identifier from the coding index, wherein the distribution identifier is configured to identify the pulse distribution, on a track, of the pulses to be encoded on the track; determining the pulse distribution, on a track, of all the pulses to be encoded on the track according to the distribution identifier; and reconstructing the pulse order on the track according to the pulse distribution.

APPARATUS AND METHOD FOR SYNTHESIZING AN AUDIO SIGNAL, DECODER, ENCODER, SYSTEM AND COMPUTER PROGRAM

A method and an apparatus for synthesizing an audio signal are described. A spectral tilt is applied to the code of a codebook used for synthesizing a current frame of the audio signal. The spectral tilt is based on the spectral tilt of the current frame of the audio signal. Further, an audio decoder operating in accordance with the inventive approach is described. 1. An apparatus for synthesizing an audio signal , comprising:an input for receiving an encoded audio signal,a decoder for decoding the encoded audio signal, the decoder comprising an adaptive codebook and a fixed codebook, and the encoded audio signal being an encoded speech signal,a filter coupled to the fixed codebook and configured to apply a spectral tilt to a code of the fixed codebook for obtaining a filtered code of the fixed codebook,a summer coupled to the adaptive codebook and to the filter, the summer configured to combine a code from the adaptive codebook and the filtered code of the fixed codebook for obtaining a combined code, anda LPC synthesis filter coupled to the summer and configured to synthesize the audio signal,wherein the spectral tilt is based on the spectral tilt of the current frame of the audio signal,wherein the apparatus is configured to determine the spectral tilt of the current frame of the audio signal on the basis of spectral envelope information for the current frame of the audio signal, andwherein the filter is configured to apply the spectral tilt by filtering the code of the fixed codebook based on a transfer function modeling the spectral tilt.4. The apparatus of claim 3 , wherein N is equal to the number of codes in the codebook.5. The apparatus of claim 1 , wherein the transfer function comprising the spectral tilt is defined as follows:{'br': None, 'i': F', 'z', 'z, 'sub': 't1', 'sup': '−1', '()=1−γ.'}6. The apparatus of claim 1 , wherein the apparatus is configured to combine the determined spectral tilt of the current frame of the audio signal with a factor ...

CODING METHOD, DECODING METHOD, CODER, AND DECODER

A coding method, a decoding method, a coder, and a decoder, where the coding method includes obtaining the pulse distribution, on a track, of the pulses to be encoded on the track, determining a distribution identifier for identifying the pulse distribution according to the pulse distribution, and generating a coding index that includes the distribution identifier. The decoding method includes receiving a coding index, obtaining a distribution identifier from the coding index, wherein the distribution identifier is configured to identify the pulse distribution, on a track, of the pulses to be encoded on the track, determining the pulse distribution, on a track, of all the pulses to be encoded on the track according to the distribution identifier, and reconstructing the pulse order on the track according to the pulse distribution. 2. The method of claim 1 , further comprising:obtaining pulse sign information indicative of positive and negative features of the pulses; anddetermining a pulse sign index corresponding to the pulse sign information after determining the third index, wherein the coding index further comprises the pulse sign index corresponding to the pulse sign information for each pulse.3. The method of claim 1 , wherein the generating the coding index comprises:overlaying information about other indices with the first index used as a start value, wherein a value of the first index corresponds to an independent value range of the coding index; and{'b': 3', '2', '2', '3, 'overlaying a combination of the second index and the third index according to I×W(N)+I, wherein the I represents the second index, wherein the I represents the third index, wherein the W(N) represents a total quantity of all possible distributions of the pulse positions on the track, and wherein the N represents the quantity of the pulse positions corresponding to the first index.'}5. The coding device of claim 4 , wherein the processor is further configured to:obtain pulse sign ...

METHOD FOR SPEECH CODING, METHOD FOR SPEECH DECODING AND THEIR APPARATUSES

A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result 12-. (canceled)3. A method comprising:decoding an excitation vector based on an excitation code, the excitation vector having a number of samples with zero amplitude;modifying the excitation vector based on a gain such that the number of samples with zero amplitude is changed; andsynthesizing a speech signal based on the modified excitation vector.4. The method of claim 3 , further comprising:obtaining an adaptive code vector from an adaptive codebook based on an adaptive code associated with received coded speech.5. The method of claim 4 , wherein the gain is decoded in a decoding period associated with the received coded speech.6. The method of claim 4 , wherein the excitation vector is modified based on a noise level associated with the received coded speech.7. The method of claim 4 , further comprising:weighting the adaptive code vector and the modified excitation vector; andadding together the weighted adaptive code vector and the weighted modified excitation vector.8. The method of claim 4 , wherein the adaptive codebook is based on a past excitation.9. The method of claim 7 , further comprising:decoding a linear prediction parameter from a linear prediction parameter code associated with the received coded speech; andwherein the speech signal is synthesized using the linear prediction parameter and the added weighted adaptive code vector and weighted modified excitation vector.10. The method of claim 9 , wherein the decoded linear prediction ...

Fast computation of excitation pattern, auditory pattern and loudness

A method includes the steps of calculating a power spectrum from an auditory stimulus, filtering the power spectrum to obtain an effective power spectrum, calculating an intensity pattern from the effective power spectrum, calculating a median intensity pattern from the intensity pattern, determining an initial set of pruned detector locations, examining the initial set of pruned detector locations to determine an enhanced set of pruned detector locations, and calculating an excitation pattern from the effective power spectrum using the enhanced set of pruned detector locations. By determining the enhanced set of pruned detector locations from the initial set of pruned detector locations and computing the excitation pattern therefrom, the computational complexity of the above method can be significantly reduced when compared to conventional approaches while maintaining the accuracy thereof.

Bandwidth Extension Audio Decoding Method and Device for Predicting Spectral Envelope

A signal decoding method and device, where the method includes decoding a bit stream of a voice signal or an audio signal to acquire a decoded signal, predicting an excitation signal of an extension band according to the decoded signal, where the extension band is adjacent to a band of the decoded signal, and the band of the decoded signal is lower than the extension band; selecting a first band and a second band from the decoded signal, and predicting a spectral envelope of the extension band according to a spectral coefficient of the first band and a spectral coefficient of the second band; and determining a frequency-domain signal of the extension band according to the spectral envelope of the extension band and the excitation signal of the extension band. 1. A signal encoding method , comprising:performing core layer encoding on at least one of a voice signal or an audio signal and obtaining a core layer bit stream of the at least one of the voice or the audio signal from the core layer encoding;performing extension layer processing on the at least one of the voice or the audio signal and determining a first envelope of an extension band according to the extension layer processing;determining a second envelope of the extension band according to a signal-to-noise ratio of the at least one of the voice or the audio signal, a pitch period of the at least one of the voice or the audio signal, and the first envelope of the extension band;encoding the second envelope and obtaining an extension layer bit stream according to the encoding of the second envelope; andsending the core layer bit stream and the extension layer bit stream to a decoder end.2. A signal decoding method , comprising:receiving, from an encoder end, a core layer bit stream and an extension layer bit stream of at least one of a voice or audio signal;decoding the extension layer bit stream and determining a second envelope of an extension band according to the decoding the extensions layer bitstream, ...

Method and apparatus for adaptively encoding and decoding high frequency band

Provided are a method and apparatus for encoding and decoding an audio signal. According to the present application, a signal of a high frequency band above a preset frequency band is adaptively encoded or decoded in the time domain or in the frequency domain by using a signal of a low frequency band below the preset frequency band. As such, the sound quality of a high frequency signal is not deteriorate even when an audio signal is encoded or decoded by using a small number of bits and thus coding efficiency may be maximized.

METHOD OF DETECTING A PREDETERMINED FREQUENCY BAND IN AN AUDIO DATA SIGNAL, DETECTION DEVICE AND COMPUTER PROGRAM CORRESPONDING THERETO

A method is provided for detecting a predetermined frequency band in an audio data signal which has previously been coded according to a succession of data blocks, among which at least certain blocks contain respectively at least one set of spectral parameters representing a linear prediction filter. Such a method of detection implements, for a current block among the at least certain blocks and for which at least a plurality of spectral parameters of the set have been previously decoded, acts of: determining, among the plurality of previously decoded spectral parameters, the index of the first spectral parameter closest to a threshold frequency; calculating at least one criterion on the basis of the determined index; and deciding whether the predetermined frequency band is detected in the current block, as a function of the criterion calculated.

Audio Decoder Having A Bandwidth Extension Module With An Energy Adjusting Module

An audio decoder configured to produce an audio signal from a bitstream containing audio frames includes: a core band decoding module configured to derive a directly decoded core band audio signal from the bitstream; a bandwidth extension module configured to derive a parametrically de-coded bandwidth extension audio signal from the core band audio signal and from the bitstream, wherein the bandwidth extension audio signal is based on a frequency domain signal having at least one frequency band; and a combiner configured to combine the core band audio signal and the bandwidth extension audio signal so as to produce the audio signal; wherein the bandwidth extension module includes an energy adjusting module being configured in such way that in a current audio frame in which an audio frame loss occurs, an adjusted signal energy for the cur-rent audio frame for the at least one frequency band is set. 1. An audio decoder configured to produce an audio signal from a bitstream comprising audio frames , the audio decoder comprising:a core band decoding module configured to derive a directly decoded core band audio signal from the bitstream;a bandwidth extension module configured to derive a parametrically decoded o bandwidth extension audio signal from the core band audio signal and from the bitstream, wherein the bandwidth extension audio signal is based on a frequency domain signal comprising at least one frequency band; anda combiner configured to combine the core band audio signal and the bandwidth extension audio signal so as to produce the audio signal;wherein the bandwidth extension module comprises an energy adjusting module being configured in such way that in a current audio frame in which an audio frame loss occurs, an adjusted signal energy for the current audio frame for the at least one frequency band is setbased on a current gain factor for the current audio frame, wherein the current gain factor is derived from a gain factor from a previous audio frame or ...

METHOD AND ELECTRONIC DEVICE FOR JOINTLY PLAYING HIGH-FIDELITY SOUNDS OF MULTIPLE PLAYERS

Disclosed are a data download method and a method and electronic device for jointly playing high-fidelity sounds of multiple players includes: receiving first sound data of a HIFI audio format and second sound data of a second audio format; transmitting the first sound data of the HIFI audio format to a first processor for decoding, decoding the second sound data of the second audio format, and converting the decoded second sound data into the second audio data of a PCM format; controlling the first processor to convert the second sound data of the PCM format into the second sound data of the HIFI audio format; and mixing and playing the first sound data and the second sound data of the HIFI audio format. The disclosure realizes reminder function while ensuring sound mixing joint playing of multiple paths of sound sources. 1. A method for jointly playing HIFI sounds of multiple players , executed by an electronic device , comprising:receiving first sound data of a HIFI audio format and second sound data of a second audio format, wherein the second audio format is an audio format except the HIFI format;transmitting the first sound data of the HIFI audio format to a first processor for decoding, decoding the second sound data of the second audio format, and converting the decoded second sound data into the second audio data of a pulse coding modulation PCM format;controlling the first processor to convert the second sound data of the PCM format into the second sound data of the HIFI audio format; andmixing and playing the first sound data and the second sound data of the HIFI audio format.2. The method according to claim 1 , wherein the first audio data of the HIFI audio format is decoded in a hardware decoding mode; and the sound data of the second audio format is decoded in a software decoding mode.3. The method according to claim 1 , wherein the PCM format is 48 k 16 bit PCM format.4. The method according to claim 1 , wherein mixing and playing the first sound data ...

ENCODING METHOD, DECODING METHOD, ENCODER, DECODER, PROGRAM AND RECORDING MEDIUM

A frequency-domain sample interval corresponding to a time-domain pitch period L corresponding to a time-domain pitch period code of an audio signal in a given time period is obtained as a converted interval T, a frequency-domain pitch period T is chosen from among candidates including the converted interval Tand integer multiples U×Tof the converted interval T, and a frequency-domain pitch period code indicating how many times the frequency-domain pitch period T is greater than the converted interval Tis obtained. The frequency-domain pitch period code is output so that a decoding side can identify the frequency-domain pitch period T. 1. An encoding method comprising:{'sub': '1', 'a period conversion step of receiving a time-domain pitch period L corresponding to a time-domain pitch period code of an audio signal in a given time period, obtaining, as a converted interval T, a sample interval in an N-points frequency-domain sample string, the sample interval corresponding to the time-domain pitch period L, and outputting the time-domain pitch period code to a decoder;'}{'sub': 1', '1', '1, 'a frequency-domain pitch period analysis step of receiving the N-points frequency-domain sample string derived from the audio signal in the given time period, choosing a first frequency-domain pitch period T from among a plurality of candidates including integer multiples U×Tof the converted interval T, where U is an integer in a predetermined first range, the first frequency-domain pitch period T being a pitch period in the N-points frequency-domain sample string derived from the audio signal, obtaining a first frequency-domain pitch period code indicating how many times the first frequency-domain pitch period T is greater than the converted interval T, and outputting the first frequency-domain pitch period code to the decoder; and'}a frequency-domain-pitch-period-based encoding step of encoding a first sample group of all or some of one or a plurality of successive samples ...

ENCODING METHOD, DECODING METHOD, ENCODER, DECODER, PROGRAM AND RECORDING MEDIUM

A frequency-domain sample interval corresponding to a time-domain pitch period L corresponding to a time-domain pitch period code of an audio signal in a given time period is obtained as a converted interval T, a frequency-domain pitch period T is chosen from among candidates including the converted interval Tand integer multiples U×Tof the converted interval T, and a frequency-domain pitch period code indicating how many times the frequency-domain pitch period T is greater than the converted interval Tis obtained. The frequency-domain pitch period code is output so that a decoding side can identify the frequency-domain pitch period T. 1. An encoding method comprising:a long-term prediction analysis step of receiving an audio signal in a given time period, performing time-domain long-term prediction analysis of the audio signal in the given time period to obtain a time-domain pitch period L and a time-domain pitch period code corresponding to the time-domain pitch period L, and outputting the time-domain pitch period code to a decoder;a long-term prediction residual generation step of using the time-domain pitch period L to obtain a long-term prediction residual signal of the audio signal;a frequency-domain sample string generation step of obtaining an N-points frequency-domain sample string which is derived from the long-term prediction residual signal or an N-points frequency-domain sample string which is derived from the audio signal;{'sub': '1', 'a period conversion step of obtaining, as a converted interval T, a sample interval in the N-points frequency-domain sample string, the sample interval corresponding to the time-domain pitch period L;'}{'sub': 1', '1', '1, 'a frequency-domain pitch period analysis step of receiving the N-points frequency-domain sample string, choosing a first frequency-domain pitch period T from among a plurality of candidates including integer multiples U×Tof the converted interval T, where U is an integer in a predetermined first ...

Device And Method For Quantizing The Gains Of The Adaptive And Fixed Contributions Of The Excitation In A Celp Codec

A device and method for quantizing a gain of a fixed contribution of an excitation in a frame, including sub-frames, of a coded sound signal, wherein the gain of the fixed excitation contribution is estimated in a sub-frame using a parameter representative of a classification of the frame. The gain of the fixed excitation contribution is then quantized in the sub-frame using the estimated gain. The device and method is used in jointly quantizing gains of adaptive and fixed contributions of an excitation in a frame of a coded sound signal. For retrieving a quantized gain of a fixed contribution of an excitation in a sub-frame of a frame, the gain of the fixed excitation contribution is estimated using a parameter representative of a classification of the frame, a gain codebook supplies a correction factor in response to a received, gain codebook index, and a multiplier multiplies the estimated gain by the correction factor to provide a quantized gain of the fixed excitation contribution. 150-. (canceled)51. A device for coding a sound signal , comprising:at least one processor; and 'a CELP coder configured to produce, in response to the sound signal, sound signal encoding parameters including (a) an adaptive codebook contribution of an excitation for a synthesis filter, (b) an adaptive codebook gain for scaling the adaptive codebook contribution, and (c) a fixed codebook contribution of the excitation; and', 'a memory coupled to the processor and comprising non-transitory code instructions that when executed cause the processor to implement the estimator is supplied with a parameter representative of a classification of the frame;', 'the estimator, for a first sub-frame of the frame, uses the parameter representative of the classification of the frame and an energy of the fixed codebook contribution to estimate the fixed codebook gain; and', 'the estimator, for each sub-frame of the frame following the first sub-frame, uses the parameter representative of the ...

TEXT CATEGORIZATION USING NATURAL LANGUAGE PROCESSING

A method performed by a device may include identifying a plurality of samples of textual content; performing tokenization of the plurality of samples to generate a respective plurality of tokenized samples; performing embedding of the plurality of tokenized samples to generate a sample matrix; determining groupings of attributes of the sample matrix using a convolutional neural network; determining context relationships between the groupings of attributes using a bidirectional long short term memory (LSTM) technique; selecting predicted labels for the plurality of samples using a model, wherein the model selects, for a particular sample of the plurality of samples, a predicted label of the predicted labels from a plurality of labels based on respective scores of the particular sample with regard to the plurality of labels and based on a nonparametric paired comparison of the respective scores; and providing information identifying the predicted labels. 1. A method , comprising: 'the plurality of samples of textual context including user-generated reviews of products;', 'identifying, by a device, a plurality of samples of textual content,'} 'the vocabulary set including company-specific terms or jargon;', 'the domain-specific corpus being based on a vocabulary set that is specific to a domain associated with the plurality of samples,'}, 'performing, by the device, tokenization of the plurality of samples of textual content to generate a respective plurality of tokenized samples using a domain-specific corpus,'}performing, by the device, embedding of the respective plurality of tokenized samples using the domain-specific corpus to generate a sample matrix; 'the groupings of attributes of the sample matrix being passed to a bidirectional long short-term memory (LSTM) layer;', 'determining, by the device, groupings of attributes of the sample matrix using a convolutional neural network,'}determining, by the device, context relationships between the groupings of ...

BANDWIDTH EXTENSION METHOD AND APPARATUS

The present invention provide a bandwidth extension method and apparatus. The method includes: acquiring a bandwidth extension parameter, where the bandwidth extension parameter includes one or more of the following parameters: a linear predictive coefficient (LPC), a line spectral frequency (LSF) parameter, a pitch period, a decoding rate, an adaptive codebook contribution, and an algebraic codebook contribution; and performing, according to the bandwidth extension parameter, bandwidth extension on a decoded low-frequency signal, to obtain a high frequency band signal. The high frequency band signal recovered by using the bandwidth extension method and apparatus in the embodiments of the present invention is close to an original high frequency band signal, and the quality is satisfactory. 1. A decoder implemented bandwidth extension method , comprising:receiving a bit stream encoded from an original signal;performing decoding operations on the bit stream, wherein a low frequency signal is generated via the decoding operations, wherein a collection of parameters is acquired via the decoding operations, and wherein the collection of parameters comprises one or more of the following parameters: a linear predictive coefficient (LPC), a line spectral frequency (LSF) parameter, a pitch period, a decoding rate, an adaptive codebook contribution signal, and an algebraic codebook contribution signal;performing, according to the collection of parameters, bandwidth extension operations on the decoded low-frequency signal, wherein a high frequency excitation signal and a high frequency energy are obtained via the bandwidth extension operations; andgenerating a high frequency band signal from the high frequency excitation signal and the high frequency energy, the high frequency band signal to recover a high frequency component of the original signal.2. The method according to claim 1 , wherein the bandwidth extension operations comprise:predicting the high-frequency energy and ...

Differential dynamic content delivery with text display

Differential dynamic content delivery including providing a session document for a presentation, wherein the session document includes a session grammar and a session structured document; selecting from the session structured document a classified structural element in dependence upon user classifications of a user participant in the presentation; presenting the selected structural element to the user; streaming presentation speech to the user including individual speech from at least one user participating in the presentation; converting the presentation speech to text; detecting whether the presentation speech contains simultaneous individual speech from two or more users; and displaying the text if the presentation speech contains simultaneous individual speech from two or more users.

Indicating frame parameter reusability for coding vectors

In general, techniques are described for indicating frame parameter reusability for decoding vectors. A device comprising a processor and a memory may perform the techniques. The processor may be configured to obtain a bitstream comprising a vector representative of an orthogonal spatial axis in a spherical harmonics domain. The bitstream may further comprise an indicator for whether to reuse, from a previous frame, at least one syntax element indicative of information used when compressing the vector. The memory may be configured to store the bitstream.

Scaling for gain shape circuitry

A method of operation of a device includes receiving a first set of samples and a second set of samples. The first set of samples corresponds to a portion of a first audio frame and the second set of samples corresponds to a second audio frame. The method further includes generating a target set of samples based on the first set of samples and a first subset of the second set of samples and generating a reference set of samples based at least partially on a second subset of the second set of samples. The method also includes scaling the target set of samples to generate a scaled target set of samples and generating a third set of samples based on the scaled target set of samples and one or more samples of the second set of samples.

Method and apparatus for predicting high band excitation signal

A method and an apparatus for predicting a high band excitation signal are disclosed. The method includes: acquiring, according to a received low band bitstream, a set of spectral frequency parameters that are arranged in an order of frequencies, calculating a spectral frequency parameter difference between every two spectral frequency parameters that have a same position interval; acquiring a minimum spectral frequency parameter difference from the calculated spectral frequency parameter differences; determining, according to a frequency bin that corresponds to the minimum spectral frequency parameter difference, a start frequency bin for predicting a high band excitation signal from a low band; and predicting the high band excitation signal from the low band according to the start frequency bin. By implementing embodiments of the present invention, a high band excitation signal can be better predicted, thereby improving performance of the high band excitation signal.

NOISE FILLING WITHOUT SIDE INFORMATION FOR CELP-LIKE CODERS

An audio decoder provides a decoded audio information on the basis of an encoded audio information including linear prediction coefficients (LPC) and includes a tilt adjuster to adjust a tilt of a noise using linear prediction coefficients of a current frame to acquire a tilt information and a noise inserter configured to add the noise to the current frame in dependence on the tilt information. Another audio decoder includes a noise level estimator to estimate a noise level for a current frame using a linear prediction coefficient of at least one previous frame to acquire a noise level information; and a noise inserter to add a noise to the current frame in dependence on the noise level information provided by the noise level estimator. Thus, side information about a background noise in the bit-stream may be omitted. Methods and computer programs serve a similar purpose. 1. An audio decoder for providing a decoded audio information on the basis of an encoded audio information comprising linear prediction coefficients (LPC) ,the audio decoder comprising:a tilt adjuster configured to adjust a tilt of a background noise, wherein the tilt adjuster is configured to use linear prediction coefficients of a current frame to acquire a tilt information; anda noise level estimator; anda decoder core configured to decode an audio information of the current frame using the linear prediction coefficients of the current frame to acquire a decoded core coder output signal; anda noise inserter configured to add the adjusted background noise to the current frame, to perform a noise filling.2. The audio decoder according to claim 1 , wherein the audio decoder comprises a frame type determinator for determining a frame type of the current frame claim 1 , the frame type determinator being configured to activate the tilt adjuster to adjust the tilt of the background noise when the frame type of the current frame is detected to be of a speech type.3. The audio decoder according to claim 1 ...

Method And Apparatus For Reconstructing Signal During Stereo Signal Encoding

Example signal reconstructing method and apparatus are described. One example method includes obtaining a reference sound channel and a target sound channel. An adaptive length of a transition segment is obtained based on an inter-channel time difference in the current frame and an initial length of the transition segment. A transition window in the current frame is obtained based on the adaptive length of the transition segment. A gain modification factor of a reconstructed signal is obtained. A transition segment signal on the target sound channel is obtained based on the inter-channel time difference, the adaptive length of the transition segment, the transition window, the gain modification factor, and a reference sound channel signal and a target sound channel signal. 1. A method for reconstructing a signal during stereo signal encoding , comprising:obtaining a reference sound channel and a target sound channel in a current frame;obtaining an adaptive length of a transition segment in the current frame based on an inter-channel time difference in the current frame and an initial length of the transition segment in the current frame;obtaining a transition window in the current frame based on the adaptive length of the transition segment in the current frame;obtaining a gain modification factor of a reconstructed signal in the current frame; andobtaining a transition segment signal on the target sound channel in the current frame based on the inter-channel time difference in the current frame, the adaptive length of the transition segment in the current frame, the transition window in the current frame, the gain modification factor in the current frame, a reference sound channel signal in the current frame, and a target sound channel signal in the current frame.2. The method according to claim 1 , wherein the obtaining an adaptive length of a transition segment in the current frame based on an inter-channel time difference in the current frame and an initial ...

BANDWIDTH EXTENSION METHOD AND APPARATUS

The present invention provide a bandwidth extension method and apparatus. The method includes: acquiring a bandwidth extension parameter, where the bandwidth extension parameter includes one or more of the following parameters: a linear predictive coefficient (LPC), a line spectral frequency (LSF) parameter, a pitch period, a decoding rate, an adaptive codebook contribution, and an algebraic codebook contribution; and performing, according to the bandwidth extension parameter, bandwidth extension on a decoded low frequency band signal, to obtain a high frequency band signal. The high frequency band signal recovered by using the bandwidth extension method and apparatus in the embodiments of the present invention is close to an original high frequency band signal, and the quality is satisfactory. 1. A decoder implemented bandwidth extension method , comprising:performing decoding operations on a bitstream encoded from an audio signal, wherein a low frequency band signal is generated via the decoding operations, wherein a collection of parameters is acquired via the decoding operations, and wherein the collection of parameters comprises one or more of the following parameters: a linear predictive coefficients (LPC), a set of line spectral frequency (LSF) parameters, a pitch period, an adaptive codebook contribution, and an algebraic codebook contribution;predicting a high frequency band gain according to the LPC, and predicting a high frequency band excitation signal by selecting a frequency band from a low frequency band excitation signal according to difference values between every two LSF parameters of the set of LSF parameters, wherein the selected frequency band corresponds to a pair of LSF parameters that have a lower difference value than every other pair of LSF parameters of the set of LSF parameters, and wherein the low frequency band excitation signal is represented by a sum of the adaptive codebook contribution and the algebraic codebook contribution; ...

Apparatus for quantizing linear predictive coding coefficients, sound encoding apparatus, apparatus for de-quantizing linear predictive coding coefficients, sound decoding apparatus, and electronic device therefore

A quantizing apparatus is provided that includes a quantization path determiner that determines a path from a first path not using inter-frame prediction and a second path using the inter-frame prediction, as a quantization path of an input signal, based on a criterion before quantization of the input signal; a first quantizer that quantizes the input signal, if the first path is determined as the quantization path of the input signal; and a second quantizer that quantizes the input signal, if the second path is determined as the quantization path of the input signal.

Audio Signal Encoding and Decoding Method, and Audio Signal Encoding and Decoding Apparatus

An audio signal encoding and decoding method, an audio signal encoding and decoding apparatus, a transmitter, a receiver, and a communications system, which can improve encoding and/or decoding performance. The audio signal encoding method includes dividing a to-be-encoded time domain signal into a low band signal and a high band signal; encoding the low band signal to obtain a low frequency encoding parameter; calculating a voiced degree factor, and predicting a high band excitation signal; weighting the high band excitation signal and random noise using the voiced degree factor, so as to obtain a synthesized excitation signal; and obtaining a high frequency encoding parameter based on the synthesized excitation signal and the high band signal. Technical solutions in the embodiments of the present invention can improve an encoding or decoding effect.

CONCEPT FOR ENCODING AN AUDIO SIGNAL AND DECODING AN AUDIO SIGNAL USING DETERMINISTIC AND NOISE LIKE INFORMATION

An encoder for encoding an audio signal has: an analyzer configured for deriving prediction coefficients and a residual signal from an unvoiced frame of the audio signal; a gain parameter calculator configured for calculating a first gain parameter information for defining a first excitation signal related to a deterministic codebook and for calculating a second gain parameter information for defining a second excitation signal related to a noise-like signal for the unvoiced frame; and a bitstream former configured for forming an output signal based on an information related to a voiced signal frame, the first gain parameter information and the second gain parameter information. 1. An encoder for encoding an audio signal , the encoder comprising:an analyzer configured for deriving prediction coefficients and a residual signal from an unvoiced frame of the audio signal;a gain parameter calculator configured for calculating a first gain parameter information for defining a first excitation signal related to a deterministic codebook and for calculating a second gain parameter information for defining a second excitation signal related to a noise-like signal for the unvoiced frame; anda bitstream former configured for forming an output signal based on an information related to a voiced signal frame, the first gain parameter information and the second gain parameter information.2. The encoder according to claim 1 , wherein the gain parameter calculator is configured for calculating a first gain parameter and a second gain parameter and wherein the bitstream former is configured for forming the output signal based on the first gain parameter and the second gain parameter; orwherein the gain parameter calculator comprises a quantizer configured for quantizing the first gain parameter for acquiring a first quantized gain parameter and for quantizing the second gain parameter for acquiring a second quantized gain parameter and wherein the bitstream former is configured for ...

Concept for encoding an audio signal and decoding an audio signal using speech related spectral shaping information

According to an aspect of the present invention an encoder for encoding an audio signal has an analyzer configured for deriving prediction coefficients and a residual signal from a frame of the audio signal. The encoder has a formant information calculator configured for calculating a speech related spectral shaping information from the prediction coefficients, a gain parameter calculator configured for calculating a gain parameter from an unvoiced residual signal and the spectral shaping information and a bitstream former configured for forming an output signal based on an information related to a voiced signal frame, the gain parameter or a quantized gain parameter and the prediction coefficients.

Audio encoder for encoding a multichannel signal and audio decoder for decoding an encoded audio signal

A schematic block diagram of an audio encoder for encoding a multichannel audio signal is shown. The audio encoder includes a linear prediction domain encoder, a frequency domain encoder, and a controller for switching between the linear prediction domain encoder and the frequency domain encoder. The controller is configured such that a portion of the multichannel signal is represented either by an encoded frame of the linear prediction domain encoder or by an encoded frame of the frequency domain encoder. The linear prediction domain encoder includes a downmixer for downmixing the multichannel signal to obtain a downmixed signal. The linear prediction domain encoder further includes a linear prediction domain core encoder for encoding the downmix signal and furthermore, the linear prediction domain encoder includes a first joint multichannel encoder for generating first multichannel information from the multichannel signal.

CONCEPT FOR ENCODING AN AUDIO SIGNAL AND DECODING AN AUDIO SIGNAL USING DETERMINISTIC AND NOISE LIKE INFORMATION

An encoder for encoding an audio signal has: an analyzer configured for deriving prediction coefficients and a residual signal from an unvoiced frame of the audio signal; a gain parameter calculator configured for calculating a first gain parameter information for defining a first excitation signal related to a deterministic codebook and for calculating a second gain parameter information for defining a second excitation signal related to a noise-like signal for the unvoiced frame; and a bitstream former configured for forming an output signal based on an information related to a voiced signal frame, the first gain parameter information and the second gain parameter information. 1. An encoder for encoding an audio signal , the encoder comprising:an analyzer configured for deriving prediction coefficients and a residual signal from an unvoiced frame of the audio signal;a gain parameter calculator configured for calculating a first gain parameter information for defining a first excitation signal related to a deterministic codebook and for calculating a second gain parameter information for defining a second excitation signal related to a noise-like signal for the unvoiced frame; anda bitstream former configured for forming an output signal based on an information related to a voiced signal frame, the first gain parameter information and the second gain parameter information.2. The encoder according to claim 1 , wherein the gain parameter calculator is configured for calculating a first gain parameter and a second gain parameter and wherein the bitstream former is configured for forming the output signal based on the first gain parameter and the second gain parameter; orwherein the gain parameter calculator comprises a quantizer configured for quantizing the first gain parameter for acquiring a first quantized gain parameter and for quantizing the second gain parameter for acquiring a second quantized gain parameter and wherein the bitstream former is configured for ...

PROCESSING METHOD OF AUDIO SIGNAL AND ELECTRONIC DEVICE SUPPORTING THE SAME

According to an embodiment, the above-described specification discloses an electronic device comprises at least one processor configured to: receive a first audio signal and a second audio signal; detect a spectral envelope signal from the first audio signal and extract a feature point from the second audio signal; extend a high-band of the second audio signal based on the spectral envelope signal from the first audio signal and the feature point from the second audio signal to generate a high-band extension signal; and mix the high-band extension signal and the first audio signal, thereby resulting in a synthesized signal. 1. An electronic device comprising:at least one processor configured to:receive a first audio signal and a second audio signal;detect a spectral envelope signal from the first audio signal and extract a feature point from the second audio signal;extend a high-band of the second audio signal based on the spectral envelope signal from the first audio signal and the feature point from the second audio signal to generate a high-band extension signal; andmix the high-band extension signal and the first audio signal, thereby resulting in a synthesized signal.2. The electronic device of claim 1 , further comprising a first microphone and a second microphone operatively connected to the at least one processor claim 1 , wherein the first microphone includes an external microphone disposed on one side of an earphone or a headset and the second microphone disposed on one side of a housing configured to be mounted in an ear.3. The electronic device of claim 2 , wherein the second microphone includes at least one of an in-ear microphone or a bone conduction microphone.4. The electronic device of claim 1 , wherein the first audio signal includes a signal in a band wider than the second audio signal.5. The electronic device of claim 1 , wherein the second audio signal includes greater energy in a low-band than the first audio signal.6. The electronic device of ...

Packet Loss Concealment for Speech Coding

A speech coding method of reducing error propagation due to voice packet loss, is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame. The method is used for a voiced speech class. A pitch cycle length is compared to a subframe size to decide to reduce the pitch gain for the first subframe or the first two subframes within the frame. A strongly voiced class is decided by checking if the pitch lags are stable and the pitch gains are high enough with the frame; for the strongly voiced frame, the pitch lags and the pitch gains can be encoded more efficiently than other speech classes.

INTER-CHANNEL ENCODING AND DECODING OF MULTIPLE HIGH-BAND AUDIO SIGNALS

A device includes an encoder and a transmitter. The encoder is configured to generate a first high-band portion of a first signal based on a left signal and a right signal. The encoder is also configured to generate a set of adjustment gain parameters based on a high-band non-reference signal. The high-band non-reference signal corresponds to one of a left high-band portion of the left signal or a right high-band portion of the right signal as a high-band non-reference signal. The transmitter is configured to transmit information corresponding to the first high-band portion of the first signal. The transmitter is also configured to transmit the set of adjustment gain parameters corresponding to the high-band non-reference signal. 1. A device comprising: generate a first high-band portion of a first signal based on a left signal and a right signal; and', 'generate a set of adjustment gain parameters based on a high-band non-reference signal, the high-band non-reference signal corresponding to one of a left high-band portion of the left signal or a right high-band portion of the right signal; and, 'an encoder configured to transmit information corresponding to the first high-band portion of the first signal; and', 'transmit the set of adjustment gain parameters., 'a transmitter configured to2. The device of claim 1 , wherein the left signal corresponds to a left channel of a received stereo signal and the right signal corresponds to a right channel of the received stereo signal claim 1 , wherein the encoder is further configured to generate the first signal based on a downmix of the left signal and the right signal claim 1 , and wherein the first signal corresponds to a mid signal claim 1 , and wherein the first high-band portion of the first signal corresponds to a high-band portion of the mid signal.3. The device of claim 1 , wherein the information includes high-band linear predictive coefficient (LPC) parameters claim 1 , a set of first high-band gain parameters ...