TWI475896B - Stereo filter with monophonic compatibility and speaker compatibility - Google Patents

Stereo filter with monophonic compatibility and speaker compatibility Download PDF

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TWI475896B
TWI475896B TW098130084A TW98130084A TWI475896B TW I475896 B TWI475896 B TW I475896B TW 098130084 A TW098130084 A TW 098130084A TW 98130084 A TW98130084 A TW 98130084A TW I475896 B TWI475896 B TW I475896B
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filter
basic
stereo
khz
pair
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TW201031234A (en
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Glenn N Dickins
David S Mcgrath
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Dolby Lab Licensing Corp
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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S7/00Indicating arrangements; Control arrangements, e.g. balance control
    • H04S7/30Control circuits for electronic adaptation of the sound field
    • H04S7/305Electronic adaptation of stereophonic audio signals to reverberation of the listening space
    • H04S7/306For headphones
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S2400/00Details of stereophonic systems covered by H04S but not provided for in its groups
    • H04S2400/01Multi-channel, i.e. more than two input channels, sound reproduction with two speakers wherein the multi-channel information is substantially preserved
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S2400/00Details of stereophonic systems covered by H04S but not provided for in its groups
    • H04S2400/03Aspects of down-mixing multi-channel audio to configurations with lower numbers of playback channels, e.g. 7.1 -> 5.1
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S2420/00Techniques used stereophonic systems covered by H04S but not provided for in its groups
    • H04S2420/01Enhancing the perception of the sound image or of the spatial distribution using head related transfer functions [HRTF's] or equivalents thereof, e.g. interaural time difference [ITD] or interaural level difference [ILD]
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S3/00Systems employing more than two channels, e.g. quadraphonic
    • H04S3/008Systems employing more than two channels, e.g. quadraphonic in which the audio signals are in digital form, i.e. employing more than two discrete digital channels

Description

單音相容性及揚聲器相容性之立體聲濾波器 Stereo filter with monophonic compatibility and speaker compatibility

本揭示內容大致上有關音頻信號之信號處理,且特別有關處理用於藉由立體聲濾波器空間化的音頻輸入,使得該輸出係可在耳機上、或單音地、或經過一組喇叭播放。 The present disclosure relates generally to signal processing of audio signals, and more particularly to processing audio input for spatialization by stereo filters such that the output can be played on the headphones, either monophonically, or through a set of speakers.

處理一組用於經過耳機播放的一或多個音頻輸入信號,使得該收聽者具有收聽來自複數位在收聽室中之預先界定位置的虛擬喇叭之聲音的印象係已知的。此處理在此中被稱為空間化及立體聲化。處理該等音頻輸入信號之濾波器在此中被稱為立體聲濾波器。如果不用於此處理,一經過耳機收聽之收聽者將具有該聲音係在該收聽者之頭部內側的印象。該等音頻輸入信號可為單一信號、用於立體聲重現之一對信號、複數環繞聲音信號,例如用於4.1環繞聲音之四音頻輸入信號、用於5.1之五音頻輸入信號、用於7.1之七音頻輸入信號等,且另外可包括用於特定位置的個別信號,特別是像聲音之來源。用於待空間化之每一音頻輸入信號有一對立體聲濾波器。用於逼真之重現,該立體聲濾波器考慮由每一個虛擬喇叭至左耳及右耳之每一個的頭部相關轉移函數(HRTFs),且進一步考慮被模擬收聽室之早期回音及回響的響應兩者。 It is known to process a set of one or more audio input signals for playback through the headset such that the listener has the ability to listen to the sound of the virtual horn from a predetermined position in the listening room. This process is referred to herein as spatialization and stereoization. The filter that processes the audio input signals is referred to herein as a stereo filter. If this is not the case, a listener listening through the headset will have the impression that the sound is inside the listener's head. The audio input signals may be a single signal, a pair of signals for stereo reproduction, a plurality of surround sound signals, such as four audio input signals for 4.1 surround sound, a fifth audio input signal for 5.1, for 7.1 Seven audio input signals, etc., and additionally may include individual signals for a particular location, particularly like sources of sound. There is a pair of stereo filters for each audio input signal to be spatialized. For realistic reproduction, the stereo filter considers the head related transfer function (HRTFs) from each of the virtual horns to each of the left and right ears, and further considers the response of the early echo and reverberation of the simulated listening room. Both.

如此,藉由立體聲濾波器預處理信號以產生一對音頻輸出信號-立體聲化信號-供經過耳機收聽係已知的。 As such, the signal is pre-processed by the stereo filter to produce a pair of audio output signals - stereo signals - known for passing through the headphone listening system.

常見之案例係吾人希望藉由電子地降低用於單音重現之信號的混音經過單一喇叭、亦即單音地收聽立體聲化信號。一範例係經過一行動裝置中之單音揚聲器收聽。亦常見之案例係吾人希望經過一對緊接隔開之揚聲器收聽此聲音。於該後一案例中,該立體聲化輸出信號亦被降低混音,但藉由音頻串音而非電子地。於兩案例中,接著被降低信號混音之立體聲化聽起來不自然,特別是聽起來具有減少清晰度及音頻透明度之反射。沒有妥協該立體聲化音頻中之空間及距離的印象係難以消除此問題。 A common case is that we want to listen to a stereo signal through a single speaker, that is, a single tone, by electronically reducing the mix of signals for monophonic reproduction. An example is heard through a monophonic speaker in a mobile device. It is also common for us to listen to this sound through a pair of speakers that are next to each other. In the latter case, the stereo output signal is also reduced in mixing, but by audio crosstalk rather than electronically. In both cases, the stereoization of the reduced signal mix then sounds unnatural, especially with a reflection that reduces clarity and audio transparency. It is difficult to eliminate this problem without compromising the impression of space and distance in the stereo audio.

概觀Overview

本發明之具體實施例包括一方法、一設備、及程式邏輯,例如在一電腦可讀取媒體中編碼之程式邏輯,當執行該程式邏輯時造成該方法之實行。一方法係處理一或多個音頻輸入信號供使用立體聲濾波器透過耳機呈現,以達成該一或多個音頻輸入之虛擬立體聲化,且當在降低混音之後單音地播放時、或當透過相當緊接地隔開之揚聲器播放時,具有該等立體聲化信號聽起來不錯之額外特性。另一方法係操作一資料處理系統,用於處理一或多對立體聲濾波器特徵,例如立體聲濾波器脈衝響應,以對應於一或多對被修改之立體聲濾波器特徵、例如被修改之立體聲濾波器脈衝響應作決定,以致當一或多個音頻輸入信號係藉由具有該一或多對被修改之立體聲濾波器特徵的個別之一或 多對立體聲濾波器立體聲化時,該等立體聲化信號達成該一或多個音頻輸入之虛擬立體聲化,而具有當在降低混音之後單音地播放時、或透過相當緊接地隔開之揚聲器播放時,具有該等立體聲化信號聽起來不錯之額外特性。 Particular embodiments of the present invention include a method, a device, and program logic, such as program logic encoded in a computer readable medium, which is executed when the program logic is executed. One method is to process one or more audio input signals for presentation through a headphone using a stereo filter to achieve virtual stereoization of the one or more audio inputs, and when played monophonically after reducing the mix, or when When played back in fairly close-separated speakers, it has the extra features that these stereo signals sound good. Another method is to operate a data processing system for processing one or more pairs of stereo filter characteristics, such as a stereo filter impulse response, to correspond to one or more pairs of modified stereo filter features, such as modified stereo filtering. The impulse response is determined such that when one or more of the audio input signals are by an individual having one or more pairs of modified stereo filter characteristics or When stereo pairs of stereo filters are achieved, the stereo signals achieve virtual stereoization of the one or more audio inputs, and have speakers that are separated by a single tone when the mix is reduced, or through a fairly closely spaced speaker. When playing, it has the extra features that these stereo signals sound good.

特別之具體實施例包括用於立體聲化一組一或多個音頻輸入信號的設備。該設備包括以一或多對基本立體聲濾波器為其特徵的一對立體聲濾波器,使一對基本立體聲濾波器用於該等音頻信號輸入之每一個。每一對基本立體聲濾波器係能藉由一基本左耳濾波器及一基本右耳濾波器所代表,且更進一步能藉由一基本和濾波器及一基本差濾波器所代表。每一濾波器係能以一個別之脈衝響應為特徵。 Particular embodiments include apparatus for stereoscopically grouping one or more audio input signals. The apparatus includes a pair of stereo filters characterized by one or more pairs of basic stereo filters, a pair of basic stereo filters being used for each of the audio signal inputs. Each pair of basic stereo filters can be represented by a basic left ear filter and a basic right ear filter, and further can be represented by a basic sum filter and a basic difference filter. Each filter system can be characterized by a separate impulse response.

至少一對基本立體聲濾波器被組構成空間化其個別之音頻信號輸入,以由一個別之虛擬喇叭位置將一直接響應合併至一收聽者,且合併早期回音及收聽室之回響的響應兩者。 At least one pair of basic stereo filters are grouped to spatialize their individual audio signal inputs to combine a direct response to a listener by a different virtual horn position, and combine the response of the early echo and the echo of the listening room. .

用於該至少一對之基本立體聲濾波器:●該基本和濾波器之時頻特徵實質上係與該基本差濾波器之時頻特徵不同,使得在所有頻率,該基本和濾波器長度顯著地小於該基本差濾波器長度、該基本左耳濾波器長度、及該基本右耳濾波器長度;及●與遍及該基本左耳濾波器長度或該基本右耳濾波器長度之頻率的變化作比較,該基本和濾波器長度橫越不同頻率顯著地變化,使該基本和濾波器長度隨著增加之頻率而減少。 a basic stereo filter for the at least one pair: the fundamental and filter time-frequency features are substantially different from the time-frequency characteristics of the basic difference filter such that at all frequencies, the base and filter lengths are significantly Less than the basic difference filter length, the basic left ear filter length, and the basic right ear filter length; and - comparing with changes in the frequency of the basic left ear filter length or the basic right ear filter length The fundamental and filter lengths vary significantly across different frequencies, causing the base and filter length to decrease with increasing frequency.

該設備產生可經過耳機或在單音混合之後單音地播放的輸出信號。 The device produces an output signal that can be played monophonically through the headset or after mixing the tones.

於一些具體實施例中,針對該基本立體聲濾波器之至少一對,遍及該基本和濾波器脈衝響應之最初時間間隔,該基本和濾波器脈衝響應之變遷至一不足道位準隨著時間之消逝以頻率相依之方式逐漸地發生。 In some embodiments, for at least one pair of the basic stereo filters, the transition of the basic and filter impulse responses to an insignificant level disappears over time during the initial time interval of the basic and filter impulse responses. It gradually occurs in a frequency-dependent manner.

用於一些具體實施例,針對該基本立體聲濾波器之至少一對,該基本和濾波器遍及該變遷時間間隔在頻率成分中由最初全帶寬減少朝向一低頻截止。例如,針對該基本立體聲濾波器之至少一對,該變遷時間間隔係使得該基本和濾波器脈衝響應由全帶寬變遷直至大約3ms(毫秒)至在大約40ms低於100Hz(赫茲)。 For some embodiments, for at least one pair of the basic stereo filters, the base and filter are turned off in the frequency component from the initial full bandwidth reduction toward a low frequency throughout the transition time interval. For example, for at least one pair of the basic stereo filters, the transition time interval is such that the base and filter impulse responses are shifted from full bandwidth up to about 3 ms (milliseconds) to less than 100 Hz (hertz) at about 40 ms.

於一些具體實施例中,針對該基本立體聲濾波器之至少一對,在高於10kHz(千赫)之高頻的基本差濾波器長度係少於40ms,在3kHz及4kHz間之頻率的基本差濾波器長度係少於100ms,且在少於2kHz之頻率,該基本差濾波器長度係少於160ms。針對一些具體實施例,在高於10kHz之高頻的基本差濾波器長度係少於20ms,在3kHz及4kHz間之頻率的基本差濾波器長度係少於60ms,且在少於2kHz之頻率,該基本差濾波器長度係少於120ms。針對一些具體實施例,在高於10kHz之高頻的基本差濾波器長度係少於10ms,在3kHz及4kHz間之頻率的基本差濾波器長度係少於40ms,且在少於2kHz之頻率,該基本差濾波器長度係少於80ms。 In some embodiments, for at least one pair of the basic stereo filters, the fundamental difference filter length at a high frequency above 10 kHz (kilohertz) is less than 40 ms, and the difference in frequency between 3 kHz and 4 kHz is substantially The filter length is less than 100 ms, and at frequencies less than 2 kHz, the basic difference filter length is less than 160 ms. For some embodiments, the fundamental difference filter length at frequencies above 10 kHz is less than 20 ms, the fundamental difference filter length at frequencies between 3 kHz and 4 kHz is less than 60 ms, and at frequencies less than 2 kHz, The basic difference filter length is less than 120 ms. For some embodiments, the fundamental difference filter length at frequencies above 10 kHz is less than 10 ms, the fundamental difference filter length at frequencies between 3 kHz and 4 kHz is less than 40 ms, and at frequencies less than 2 kHz, The basic difference filter length is less than 80 ms.

於一些具體實施例中,針對該基本立體聲濾波器之至少一對,該基本差濾波器長度係少於大約800ms。於一些具體實施例中,該基本差濾波器長度係少於大約400ms。於一些具體實施例中,該基本差濾波器長度係少於大約200ms。 In some embodiments, the base difference filter length is less than about 800 ms for at least one pair of the basic stereo filters. In some embodiments, the basic difference filter length is less than about 400 ms. In some embodiments, the basic difference filter length is less than about 200 ms.

於一些具體實施例中,針對該基本立體聲濾波器之至少一對,該基本和濾波器長度隨著增加之頻率而減少,對於所有少於100Hz之頻率,該基本和濾波器長度係至少40ms及最多160ms,對於所有在100Hz及1kHz間之頻率,該基本和濾波器長度係至少20ms及最多80ms,對於所有在1kHz及2kHz間之頻率,該基本和濾波器長度係至少10ms及最多20ms,且對於所有在2kHz及20kHz間之頻率,該基本和濾波器長度係至少5ms及最多20ms。於一些具體實施例中,對於所有少於100Hz之頻率,該基本和濾波器長度係至少60ms及最多120ms,對於所有在100Hz及1kHz間之頻率,該基本和濾波器長度係至少30ms及最多60ms,對於所有在1kHz及2kHz間之頻率,該基本和濾波器長度係至少15ms及最多30ms,且對於所有在2kHz及20kHz間之頻率,該基本和濾波器長度係至少7ms及最多15ms。再者,於一些具體實施例中,對於所有少於100Hz之頻率,該基本和濾波器長度係至少70ms及最多90ms,對於所有在100Hz及1kHz間之頻率,該基本和濾波器長度係至少35ms及最多50ms,對於所有在1kHz及2kHz間之頻率,該基本和濾波器長度係至 少18ms及最多25ms,且對於所有在2kHz及20kHz間之頻率,該基本和濾波器長度係至少8ms及最多12ms。 In some embodiments, for at least one pair of the basic stereo filters, the base and filter lengths decrease with increasing frequency, and for all frequencies less than 100 Hz, the base and filter lengths are at least 40 ms and Up to 160 ms, the base and filter lengths are at least 20 ms and at most 80 ms for all frequencies between 100 Hz and 1 kHz, and for all frequencies between 1 kHz and 2 kHz, the basic and filter lengths are at least 10 ms and at most 20 ms, and The base and filter lengths are at least 5 ms and at most 20 ms for all frequencies between 2 kHz and 20 kHz. In some embodiments, the base and filter lengths are at least 60 ms and at most 120 ms for all frequencies less than 100 Hz, and the base and filter lengths are at least 30 ms and at most 60 ms for all frequencies between 100 Hz and 1 kHz. For all frequencies between 1 kHz and 2 kHz, the base and filter lengths are at least 15 ms and at most 30 ms, and for all frequencies between 2 kHz and 20 kHz, the base and filter lengths are at least 7 ms and at most 15 ms. Moreover, in some embodiments, the base and filter lengths are at least 70 ms and at most 90 ms for all frequencies less than 100 Hz, and the base and filter lengths are at least 35 ms for all frequencies between 100 Hz and 1 kHz. And up to 50ms, for all frequencies between 1kHz and 2kHz, the basic and filter lengths are Less than 18ms and up to 25ms, and for all frequencies between 2kHz and 20kHz, the basic and filter length is at least 8ms and at most 12ms.

於一些具體實施例中,針對該基本立體聲濾波器之該至少一對,該等基本立體聲濾波器特徵係由一對待匹配立體聲濾波器特徵所決定。用於一些此等具體實施例,針對該基本立體聲濾波器之該至少一對,該基本差濾波器脈衝響應實質上係在晚些時候與該待匹配立體聲濾波器之差濾波器成比例。譬如,該基本差濾波器脈衝響應在40ms之後實質上變得與該待匹配立體聲濾波器之差濾波器成比例。 In some embodiments, for the at least one pair of the basic stereo filters, the basic stereo filter characteristics are determined by a feature to be matched to the stereo filter. For some such embodiments, for the at least one pair of the basic stereo filter, the basic difference filter impulse response is substantially proportional to the difference filter of the stereo filter to be matched at a later time. For example, the basic difference filter impulse response becomes substantially proportional to the difference filter of the stereo filter to be matched after 40 ms.

特別之具體實施例包括立體聲化一組一或多個音頻輸入信號之方法。該方法包括:藉由立體聲化器過濾該組音頻輸入信號,該立體聲化器以一或多對基本立體聲濾波器為其特徵。於不同具體實施例中,該等基本立體聲濾波器係如上面在此敘述特別設備具體實施例中之概觀段落中所敘述者。 Particular embodiments include a method of stereoscopically grouping one or more audio input signals. The method includes filtering the set of audio input signals by a stereoizer characterized by one or more pairs of basic stereo filters. In various embodiments, the basic stereo filters are as described above in the overview section of the particular apparatus embodiment.

特別之具體實施例包括一操作信號處理設備之方法。該方法包括:接收一對信號,該等信號代表被組構成立體聲化一音頻信號的對應待匹配立體聲濾波器對之脈衝響應;與藉由一對濾波器處理該對被接收之信號,每一濾波器係以具有時變濾波器特徵之修改濾波器為其特徵。該處理形成一對代表對應之修改立體聲濾波器對的脈衝響應之被修改信號。該等被修改之立體聲濾波器被組構成立體聲化一音頻信號,且另具有單音混合中之低感知回響下降、與 遍及耳機的立體聲濾波器上之最小衝擊的特性。 Particular embodiments include a method of operating a signal processing device. The method includes receiving a pair of signals representing impulse responses of pairs of stereo signals to be matched that are stereoscopically formed into an audio signal, and processing the pair of received signals by a pair of filters, each The filter is characterized by a modified filter with time varying filter characteristics. The process forms a pair of modified signals representative of the corresponding impulse response of the modified stereo filter pair. The modified stereo filters are grouped to form a stereo-audio signal, and the other has a low-perceive reverberation drop in the monophonic mixture, and The minimum impact characteristic across the stereo filter of the headphones.

於一些具體實施例中,該等被修改之立體聲濾波器係以一被修改之和濾波器及一被修改之差濾波器為其特徵。該等時變濾波器被組構,使得被修改之立體聲濾波器脈衝響應包括一藉由頭部相關轉移函數所界定之直接部份,用於收聽者在一預先確定位置收聽一虛擬之喇叭。再者,與該被修改之差濾波器作比較,該被修改之和濾波器具有一顯著地減少之位準及一顯著地較短之回響時間,且由該和濾波器之脈衝響應的直接部份至該和濾波器之可忽略的響應部份有一平順之變遷,使平順之變遷係隨著時間之消逝所選擇的頻率。 In some embodiments, the modified stereo filters are characterized by a modified sum filter and a modified difference filter. The time varying filters are configured such that the modified stereo filter impulse response includes a direct portion defined by a head related transfer function for the listener to listen to a virtual horn at a predetermined location. Furthermore, in comparison with the modified difference filter, the modified sum filter has a significantly reduced level and a significantly shorter reverberation time, and the direct portion of the impulse response of the sum filter The negligible response portion of the sum filter to the filter has a smooth transition that causes the smooth transition to be selected over time.

於不同具體實施例中,該等被修改之立體聲濾波器具有上面在此用於特別設備具體實施例之概觀段落中所敘述之基本立體聲濾波器的性質。 In various embodiments, the modified stereo filters have the properties of the basic stereo filters described above in the overview section of the particular apparatus embodiment.

特別之具體實施例包括一操作信號處理設備之方法。該方法包括接收代表對應於左耳及右耳立體聲濾波器之脈衝響應的左耳信號及右耳信號,該等立體聲濾波器被組構成立體聲化一音頻信號。該方法另包括混洗該左耳信號及右耳信號,以形成一與該左及右耳信號之和成比例的和信號、及一與該左耳信號及該右耳信號間之差成比例的差信號。該方法另包括藉由一具有時變濾波器特徵之和濾波器過濾該和信號,該過濾形成一被過濾之和信號;與藉由一以該和濾波器為其特徵之差濾波器處理該差信號,該處理形成一被過濾之差信號。該方法另包括解混洗該被過濾之 和信號及該被過濾之差信號,以形成代表對應於左耳及右耳被修改的立體聲濾波器之脈衝響應的被修改之左耳信號及被修改之右耳信號。該等被修改之立體聲濾波器被組構成立體聲化一音頻信號,可藉由一被修改之和濾波器及一被修改之差濾波器所代表。於不同之具體實施例中,該等被修改之立體聲濾波器具有上面在此用於特別設備具體實施例之概觀段落中所敘述之基本立體聲濾波器的性質。 Particular embodiments include a method of operating a signal processing device. The method includes receiving a left ear signal and a right ear signal representative of an impulse response corresponding to a left ear and a right ear stereo filter, the stereo filters being grouped to form a stereoned audio signal. The method further includes shuffling the left ear signal and the right ear signal to form a sum signal proportional to the sum of the left and right ear signals, and a ratio proportional to a difference between the left ear signal and the right ear signal The difference signal. The method further includes filtering the sum signal by a sum filter having a time varying filter characteristic, the filtering forming a filtered sum signal; and processing the difference signal by a difference filter characterized by the sum filter The difference signal, the process forms a filtered difference signal. The method further comprises deblending the filtered And the signal and the filtered difference signal to form a modified left ear signal and a modified right ear signal representative of an impulse response of the stereo filter modified for the left and right ears. The modified stereo filters are grouped to form a stereo-audio signal, which can be represented by a modified sum filter and a modified difference filter. In various embodiments, the modified stereo filters have the properties of the basic stereo filter described above in the overview section for a particular apparatus embodiment.

特別之具體實施例包括,其當藉由一處理系統之至少一處理器所執行時,造成實行上面在此用於特別設備具體實施例之概觀段落中所敘述之方法具體實施例的任一個。 Particular embodiments include, when executed by at least one processor of a processing system, causing any of the specific embodiments of the method described above in the detailed paragraphs of the specific apparatus embodiments herein.

特別之具體實施例包括一電腦可讀取媒體,在其中具有程式邏輯,當藉由一處理系統之至少一處理器執行該程式邏輯時,造成實行上面在此用於特別設備具體實施例之概觀段落中所敘述之方法具體實施例的任一個。 Particular embodiments include a computer readable medium having program logic therein that, when executed by at least one processor of a processing system, results in an implementation of the specific apparatus described herein for a particular apparatus Any of the specific embodiments of the method described in the paragraph.

特別之具體實施例包括一設備。該設備包括一處理系統,其具有至少一處理器,及一儲存裝置。該儲存裝置被組構成具有程式邏輯,當執行該程式邏輯時,造成該設備實行上面在此用於特別設備具體實施例之概觀段落中所敘述之方法具體實施例的任一個。 A particular embodiment includes a device. The apparatus includes a processing system having at least one processor and a storage device. The storage device is organized into program logic that, when executed, causes the device to perform any of the specific embodiments of the method described above in the detailed section of the particular device embodiment.

特別之具體實施例可提供這些態樣、特色、或優點之所有、一些、或無任一個。特別之具體實施例可提供一或多個其他態樣、特色、或優點,由在此中之圖面、敘述、及申請專利範圍,其他態樣、特色、或優點的一或多個對於熟諳此技藝者可變得輕易明顯的。 Particular embodiments may provide all, some, or none of these aspects, features, or advantages. The specific embodiments may provide one or more other aspects, features, or advantages, and one or more of the other aspects, features, or advantages of the drawings, the description, and the claims. This artist can become easily apparent.

立體聲濾波器及記號Stereo filter and mark

圖1顯示包括一對用於處理單一輸入信號之立體聲濾波器103、104的立體聲化器101之簡化方塊圖。雖然立體聲濾波器在該技藝中大致上係已知的,包括在此中所敘述之單音播放特色的立體聲濾波器不是先前技藝。 1 shows a simplified block diagram of a stereoizer 101 that includes a pair of stereo filters 103, 104 for processing a single input signal. While stereo filters are generally known in the art, stereo filters including the monophonic playback features described herein are not prior art.

為持續此敘述,一些記號被導入。用於說明之簡潔,該等信號在此中被呈現為連續之時間函數。然而,對於任何熟諳信號處理之領域者應為明顯的是該框架同樣很好地應用於離散之時間信號,亦即,應用於已被適當地取樣及量化的信號。此等信號典型係以代表時間中之被取樣瞬時的積分為指標。卷積積分變成卷積和等。再者,那些熟諳該技藝者將了解所敘述之濾波器可於該時域或該頻域的其中之一中提供,或甚至兩者的一組合,且進一步可被提供當作有限脈衝響應FIR實施、遞迴無限脈衝響應(IIR)近似值、時間延遲等。那些細節在該敘述被刪除。 To continue this narrative, some tokens are imported. For simplicity of explanation, the signals are presented here as a continuous time function. However, it should be apparent to those skilled in the art of signal processing that the framework is equally well applied to discrete time signals, i.e., to signals that have been properly sampled and quantized. These signals are typically characterized by an integral representing the instantaneous moment of sampling in time. Convolution integrals become convolutions and so on. Moreover, those skilled in the art will appreciate that the described filter can be provided in one of the time domains or the frequency domain, or a combination of both, and can further be provided as a finite impulse response FIR. Implement and recursive infinite impulse response (IIR) approximations, time delays, etc. Those details were removed in the narrative.

再者,雖然所敘述之方法大致上係可適用於任何數目之輸入來源信號及對於任何數目之輸入來源信號輕易地一般化。亦應注意的是此敘述及公式化對於個體的頭部相關轉移函數之任何特定組不是特別的,或對於任何特別之合成或一般之頭部關係轉移函數不是特別的。該技術可被應用於任何想要之立體聲響應。 Moreover, although the methods described are generally applicable to any number of input source signals and are readily generalized for any number of input source signals. It should also be noted that this description and formulation is not specific to any particular group of individual head related transfer functions, or is not specific to any particular synthetic or general head relationship transfer function. This technique can be applied to any desired stereo response.

參考圖1,藉由u(t)標示待藉由該立體聲化器101立 體聲化之單一音頻信號,用於經過耳機105立體聲呈現,且藉由hL(t)及hR(t)分別標示該等立體聲濾波器脈衝響應,該等脈衝響應分別用於該左及右耳,而用於一收聽室中之收聽者107。該立體聲化器被設計成提供至該收聽者105收聽來自一來源-在預先界定位置之“虛擬喇叭109”的信號u(t)之聲音的感覺。 Referring to FIG. 1, a single audio signal to be stereographed by the stereoizer 101 is indicated by u(t) for stereo presentation via the headphone 105, and is indicated by h L (t) and h R (t), respectively. The stereo filter impulse responses are used for the left and right ears, respectively, and for the listener 107 in a listening room. The stereoizer is designed to provide a sensation to the listener 105 to listen to the sound of the signal u(t) from a source - "virtual horn 109" at a predefined location.

關於立體聲濾波器之設計、近似法及實施有一顯著數量的先前技藝,以藉由該等立體聲濾波器103及104之合適設計達成來源之此虛擬空間定位。該等濾波器考慮每一個耳朵之頭部相關轉移函數(HRTF),好像該喇叭109係在一完美之無回聲房間中,亦即,考慮直接由該虛擬喇叭109收聽之空間尺寸及另考慮該收聽環境中之早期反射及回響兩者。用於如何設計一些立體聲濾波器之更多細節,譬如看已發表為世界專利第WO 9914983號之國際專利申請案第PCT/AU98/00769號,且其標題為“立體聲耳機裝置中之濾波效應的利用率”;及已發表為世界專利第WO 9949574號之國際專利申請案第PCT/AU99/00002號,且其標題為“音頻信號處理方法與設備”。這些申請案之每一個指定美國。公告WO 9914983及WO 9949574之每一個的內容係以引用的方式併入本文中。 There has been a significant amount of prior art in the design, approximation, and implementation of stereo filters to achieve this virtual spatial location by the appropriate design of the stereo filters 103 and 104. The filters consider the head related transfer function (HRTF) of each ear as if the speaker 109 is in a perfect echo-free room, i.e., considering the size of the space directly listened to by the virtual speaker 109 and Listen to both early reflections and reverberations in the environment. For more details on how to design some of the stereo filters, see, for example, International Patent Application No. PCT/AU98/00769, which is hereby incorporated by reference. And the international patent application No. PCT/AU99/00002, which is entitled "Audio Signal Processing Method and Apparatus". Each of these applications is designated in the United States. The contents of each of the publications WO 9914983 and WO 9949574 are incorporated herein by reference.

如此,已被立體聲化用於耳機使用之信號可為可用的。該等信號之立體聲化處理可為一或多個預先界定立體聲濾波器,提供該等預先界定立體聲濾波器,以致一收聽者具有收聽不同型式房間中之內容的感覺。一商業之立體聲 化係已知為於杜比耳機(TM)。杜比耳機立體聲化中之立體聲濾波器對具有個別之脈衝響應,該等脈衝響應具有一共同之非空間反射結尾。再者,一些杜比耳機實施僅只提供敘述單一型式收聽室之單一組立體聲濾波器,而其他杜比耳機實施能使用標示為DH1、DH2及DH3的三組不同立體聲濾波器之一立體聲化。這些立體聲濾波器具有以下之性質: As such, signals that have been stereotyped for use with headphones can be made available. The stereo processing of the signals may be one or more pre-defined stereo filters that provide the pre-defined stereo filters such that a listener has the sensation of listening to content in different types of rooms. a commercial stereo The chemical system is known as Dolby Headphones (TM). The stereo filter pairs in Dolby Headphone Stereo have individual impulse responses with a common non-spatial reflection end. Furthermore, some Dolby Headphone implementations only provide a single set of stereo filters that describe a single type of listening room, while other Dolby Headphone implementations can stereoize using one of three different stereo filters labeled DH1, DH2, and DH3. These stereo filters have the following properties:

●DH1提供於一小、良好阻抑房間中收聽之感覺,而適當用於電影及僅只音樂之錄音。 ● DH1 is provided in a small, well-suppressed room to listen to the feeling, and is suitable for film and music only recording.

˙DH2提供於一更具音響混響房間中收聽之感覺,而特別適合於音樂收聽。 ̇DH2 provides a listening experience in a more acoustic reverberant room, and is especially suitable for music listening.

˙DH3提供於一較大房間、更像一音樂廳或一電影院中收聽之感覺。 ̇DH3 provides a feeling of listening to a larger room, more like a concert hall or a movie theater.

將該卷積操作標示為,亦即,a(t)及b(t)之卷積被標示為 在此該時間相依係不明確地顯示在左手側上,但將暗指一字母之使用。非時間相依之數量將被清楚地指示。 Mark the convolution operation as , that is, the convolution of a(t) and b(t) is marked as Here, the time dependency is not explicitly displayed on the left hand side, but will imply the use of one letter. The number of non-time dependent will be clearly indicated.

一立體聲輸出包括一標示為νL(t)之左輸出信號及一標示為νR(t)之右耳信號。該立體聲輸出係藉由以該等立體聲濾波器103、104之左及右脈衝響應卷積該來源信號u(t)所產生:左輸出信號 (1) A stereo output includes a left output signal labeled ν L (t) and a right ear signal labeled ν R (t). The stereo output is generated by convolving the source signal u(t) with the left and right impulse responses of the stereo filters 103, 104: Left output signal (1)

右輸出信號 (2) Right output signal (2)

圖1顯示單一輸入音頻信號。圖2顯示具有標示為u1(t)、u2(t)、...uM(t)之一或多個音頻輸入信號的立體聲化器之簡化方塊圖,在此M係輸入音頻信號之數目。M可為1,或超過1。用於立體重現,M=2,且更大值用於環繞立體聲信號,例如M=4用於4.1環繞立體聲,M=5用於5.1環繞立體聲,M=7用於7.1環繞立體聲等。其亦可具有多數來源,例如用於一般背景之複數輸入,加上一或多個輸入,以定位特別之來源、諸如人們在一環境中說話。有用於待空間化之每一音頻輸入信號的一對立體聲濾波器。用於逼真之重現,該等立體聲濾波器考慮用於每一虛擬喇叭位置及左與右耳之個別頭部相關轉移函數(HRTF),且進一步考慮所模擬之收聽室的早期回音及反射響。用於所示立體聲化器之左及右立體聲濾波器包括左及右耳立體聲化器,每一立體聲化器203-1及204-1、203-2及204-2、...、203-M及204-M分別具有脈衝響應h1L(t)及h1R(t)、h2L(t)及h2R(t)、...、hML(t)及hMR(t)。該左耳及右耳輸出係藉由加法器205及206所加入,以產生輸出νL(t)及νR(t)。 Figure 1 shows a single input audio signal. Figure 2 shows a simplified block diagram of a stereoizer with one or more audio input signals labeled u 1 (t), u 2 (t), ... u M (t), where the M system inputs audio signals The number. M can be 1, or exceed 1. For stereo reproduction, M=2, and larger values for surround sound signals, such as M=4 for 4.1 surround sound, M=5 for 5.1 surround sound, M=7 for 7.1 surround sound, etc. It can also have many sources, such as multiple inputs for a general background, plus one or more inputs to locate a particular source, such as a person speaking in an environment. There is a pair of stereo filters for each audio input signal to be spatialized. For realistic reproduction, these stereo filters consider the individual head related transfer function (HRTF) for each virtual horn position and the left and right ears, and further consider the early echo and reflection of the simulated listening room. . The left and right stereo filters for the stereoizer shown include left and right ear stereos, each of the stereos 203-1 and 204-1, 203-2 and 204-2, ..., 203- M and 204-M have impulse responses h 1L (t) and h 1R (t), h 2L (t), and h 2R (t), ..., h ML (t) and h MR (t), respectively. The left and right ear outputs are added by adders 205 and 206 to produce outputs ν L (t) and ν R (t).

虛擬喇叭之數目被標示為Mν。此等喇叭係在圖2中之Mν個別位置顯示為喇叭209-1、209-2、...、209-Mν。雖然典型M=Mν,這是不需要的。譬如,上混可被併入空間化一對立體輸入信號,以在耳機上發聲至該收聽者,好像有五個虛擬揚聲器。 The number of virtual speakers is labeled as M ν . These horns are shown as 209-9, 209-2, ..., 209-M ν in the individual positions of M ν in Fig. 2 . Although typical M = M ν , this is not required. For example, upmixing can be incorporated into a spatialized pair of stereo input signals to sound on the earphone to the listener as if there were five virtual speakers.

於在此中之敘述中,討論具有單一對立體聲濾波器之 特性的操作及單一對立體聲濾波器之特性。那些熟諳該技藝者將了解此等具有該對立體聲濾波器之特性的操作及該對立體聲濾波器之特性應用於諸如圖2所示組構中之每一對立體聲濾波器。 In the description herein, the discussion has a single pair of stereo filters. Features of the operation and the characteristics of a single pair of stereo filters. Those skilled in the art will appreciate that such operations with the characteristics of the pair of stereo filters and the characteristics of the pair of stereo filters are applied to each pair of stereo filters, such as the one shown in FIG.

圖3顯示一立體聲化器303之簡化方塊圖,該立體聲化器具有一或多個音頻輸入信號及產生一左輸出信號νL(t)及一標示為νR(t)之右耳信號。將藉由下混頻器305所獲得之左及右輸出信號的單音混合標示為νM(t),該下混頻器在左及右信號νL(t)之每一個及一標示為νR(t)之右耳信號上實行一些濾波,與加入、亦即混合該等經濾波之信號。隨後之敘述假設單一輸入u(t)。分別在該下混頻器305左及右輸出信號上之濾波器307及308的脈衝響應標示為mL(t)及mR(t)。隨後之敘述假設單一輸入u(t)。對於每一個此輸入發生類似之操作。該單音混合接著為 3 shows a simplified block diagram of a stereoizer 303 having one or more audio input signals and generating a left output signal ν L (t) and a right ear signal labeled ν R (t). The monophonic mixture of the left and right output signals obtained by the down mixer 305 is denoted by ν M (t), and the down mixer is labeled as each of the left and right signals ν L (t) Some filtering is performed on the right ear signal of ν R (t), and the filtered signals are added, that is, mixed. The subsequent description assumes a single input u(t). The impulse responses of filters 307 and 308 on the left and right output signals of the down mixer 305 are labeled m L (t) and m R (t), respectively. The subsequent description assumes a single input u(t). A similar operation occurs for each of these inputs. The tone mix is followed by

用於理想之單音相容性,該單音混合係與該最初之信號u(t)相同(或與其成比例)是想要的。亦即,該νM(t)=αu(t),在此α係一些比例因數常數。用於應用此,假設α=1,以下之恆等式將理想地需要應用: 在此δ(t)係該單一積分、亦被稱為所界定之狄拉克三角函數,使得該。於離散之處理中,該想要之結果係每一脈衝響應係一離散函數-係與一單位脈衝響應成比例。當然,於一實用之實施中,該等計算需要 時間,故以實際造成之濾波器施行,用於“完美”單音相容性之需求係為該單位脈衝的一時間延遲及按比例變化的版本。 For ideal monophonic compatibility, the monophonic mixture is the same (or proportional to) the original signal u(t) as desired. That is, the ν M (t) = αu(t), where α is some proportional factor constant. To apply this, assuming α = 1, the following identities would ideally require an application: Here δ(t) is the single integral, also known as the defined Dirac trigonometric function, such that . In the discrete process, the desired result is Each impulse response is a discrete function - proportional to a unit impulse response. Of course, in a practical implementation, these calculations take time, so the actual filter implementation is used for the requirement of "perfect" monophonic compatibility. A time delay and a scaled version of the pulse for that unit.

用於簡單之單音混合,mL(t)=mR(t)=δ(t)。亦即,νMLR=(hL+hR)u。故用於簡單之單音混合,理想上,用於該等立體聲化輸出的單音混合之完美重現,h L (t)+h R (t)=δ(t) (5) For simple mono mixing, m L (t) = m R (t) = δ(t). That is, ν MLR =(h L +h R ) u. Therefore, it is used for simple monophonic mixing, ideally, for the perfect reproduction of the monophonic mixture for these stereo outputs, h L ( t )+ h R ( t )= δ ( t ) (5)

其想要的是該hL(t)及hR(t)提供良好之立體聲化,亦即,該等輸出經由耳機之呈現聽起來自然的,好像該聲音係來自該虛擬喇叭位置及於一真實之收聽室中。其進一步想要的是當呈現時,該立體聲輸出之單音混合聽起來像該音頻輸入u(t)。 What it wants is that the h L (t) and h R (t) provide good stereo, that is, the output sounds natural through the presentation of the earphone, as if the sound is from the virtual horn position and In the real listening room. It is further desirable that the monophonic mix of the stereo output sounds like the audio input u(t) when presented.

那些熟諳音頻信號處理之技藝者將熟悉藉由首先實行該左及右立體聲信號之混洗以產生一和頻道及一差頻道,在一組立體信號上表達立體聲濾波操作。 Those skilled in the art of audio signal processing will be familiar with the stereo filtering operation on a set of stereo signals by first performing a shuffling of the left and right stereo signals to produce a sum channel and a difference channel.

理想上,用於一左輸入及一右立體或立體聲輸入uL(t)及uR(t),標示為uS(t)及uD(t)之和及差信號: Ideally, for a left input and a right stereo or stereo input u L (t) and u R (t), labeled as the sum of u S (t) and u D (t) and the difference signal:

該倒轉關係亦藉由一混洗操作所進行: The reverse relationship is also performed by a shuffling operation:

以混洗,該立體聲濾波器脈衝響應能被表達為一具有標示為hS(t)之脈衝響應的和濾波器,且一具有標示為hD(t)之脈衝響應的差濾波器產生分別標示為νs(t)及νD(t)之經立體聲濾波的和及差信號,以致 With shuffling, the stereo filter impulse response can be expressed as a sum filter having an impulse response labeled h S (t), and a difference filter having an impulse response labeled h D (t) produces Stereo-filtered sum and difference signals labeled ν s (t) and ν D (t) and

在此 here

該左耳及右耳立體聲濾波器脈衝響應間之倒轉關係係亦藉由一混洗操作所進行: The inverse relationship between the left ear and right ear stereo filter impulse responses is also performed by a shuffling operation:

於此敘述中,有關該左及右耳立體聲濾波器hL(t)及hR(t),討論具有脈衝響應hS(t)之和濾波器與具有脈衝響應hD(t)的差濾波器之特性。這些和及差濾波器被界定用於每一對立體聲濾波器。上面所討論之立體聲輸入純粹地用於說明。當然,該和及差濾波器之存在不會視有立體或任何特別數目之輸入而定。一和及差濾波器被界定用於每一對立體聲濾波器。 In this description, regarding the left and right ear stereo filters h L (t) and h R (t), the difference between the sum of the impulse response h S (t) and the impulse response h D (t) is discussed. The characteristics of the filter. These sum and difference filters are defined for each pair of stereo filters. The stereo input discussed above is purely illustrative. Of course, the presence of the sum and difference filters will not depend on stereo or any particular number of inputs. A sum and difference filter is defined for each pair of stereo filters.

圖4A顯示藉由混洗器(shuffler)401在一左耳立體信號uL(t)及一右耳立體信號uR(t)上之混洗操作的簡化方塊 圖,隨後有分別具有和濾波器脈衝響應及差濾波器脈衝響應hS(t)及hD(t)之和濾波器403及差濾波器404,隨後有一解混洗器405,其本質上係每一信號的一混洗器及一二等分器(halver),以產生一左耳立體聲信號輸出νL(t)及一右耳立體聲信號輸出νR(t)。 4A shows a simplified block diagram of a shuffling operation on a left ear stereo signal u L (t) and a right ear stereo signal u R (t) by a shuffler 401, followed by respectively having and filtering The impulse response of the impulse response and the differential filter impulse response h S (t) and h D (t), the filter 403 and the difference filter 404, followed by a de-shuffler 405, which is essentially a shuffling of each signal And a halver to generate a left ear stereo signal output ν L (t) and a right ear stereo signal output ν R (t).

因為脈衝響應係時間信號--對一單位脈衝輸入之響應--濾波及其他信號處理操作係可正像任何其他信號在它們上施行的。圖4B顯示藉由在一左耳立體聲濾波器脈衝響應hL(t)及一右耳立體聲濾波器脈衝響應hR(t)上之混洗器401混洗操作的簡化方塊圖,以產生該和濾波器立體聲脈衝響應hS(t)及該差濾波器立體聲脈衝響應hD(t)。亦顯示者係藉由該解混洗器405解混洗,該解混洗器405本質上係一混洗器及一二等分器,以回到該左耳立體聲濾波器脈衝響應hL(t)及該右耳立體聲濾波器脈衝響應hR(t)。 Because the impulse response is a time signal - the response to a unit of pulse input - filtering and other signal processing operations can be performed just like any other signal on them. Figure 4B shows a simplified block diagram of the shuffler 401 shuffling operation on a left ear stereo filter impulse response h L (t) and a right ear stereo filter impulse response h R (t) to produce the And the filter stereo impulse response h S (t) and the difference filter stereo impulse response h D (t). Also shown is the deshuffler by the de-mixer 405, which is essentially a shuffler and a two-part equalizer to return to the left-ear stereo filter impulse response h L ( t) and the right ear stereo filter impulse response h R (t).

注意因為線性,通常實際上,該因數被該混洗刪掉,且2之比例因數被加至該解混洗輸出,以致於一些具體實施例中:u S (t)=u L (t)+u R (t) u D (t)=u L (t)-u R (t) (8b)及 Note that because of linearity, usually, actually The factor is removed by the shuffling, and a scaling factor of 2 is added to the deshuffling output such that in some embodiments: u S ( t ) = u L ( t ) + u R ( t ) u D ( t )= u L ( t )- u R ( t ) (8b) and

因此,於在此中之敘述中,所有數量可被適當地按比例變化,如對於那些熟諳該技藝者將為清楚的。 Therefore, in the description herein, all numbers may be appropriately scaled, as will be apparent to those skilled in the art.

設計該等立體聲濾波器Designing these stereo filters

本發明之特別具體實施例包括操作一信號處理設備之方法,以修改所提供之一對立體聲濾波器特徵,以決定一對被修改之立體聲濾波器特徵。該方法的一具體實施例包括接收代表一對應對立體聲濾波器的脈衝響應之一對信號,該對應對立體聲濾波器被組構成立體聲化一音頻信號。該方法另包括藉由一對濾波器處理該對所接收之信號,該對濾波器之每一個以一具有時變濾波器特徵的被修改之濾波器為其特徵,該處理形成代表一對應對被修改之立體聲濾波器的脈衝響應之一對被修改的信號。該等被修改之立體聲濾波器被組構成將一音頻信號立體聲化至一對被立體聲化之信號,且進一步具有該被立體聲化信號之單音混合對於一收聽者聽起來自然之特性。 A particular embodiment of the invention includes a method of operating a signal processing device to modify one of the pair of stereo filter features to determine a pair of modified stereo filter features. A specific embodiment of the method includes receiving a pair of signals representative of a pair of impulse responses to a stereo filter, the corresponding pair of stereo filters being grouped to form a stereo-audio signal. The method further includes processing the pair of received signals by a pair of filters each characterized by a modified filter having time varying filter characteristics, the processing forming a pair of responses One of the impulse responses of the modified stereo filter is the modified signal. The modified stereo filters are grouped to stereo an audio signal to a pair of stereo signals, and further have the characteristics that the monophonic mixing of the stereo signal sounds natural to a listener.

考慮一組分別具有左耳及右耳脈衝響應hL(t)及hR(t)之立體聲濾波器。如上面所述,用於如方程式(3)中所敘述之單音混合,用於理想之完美單音相容性,以下之恆等式將理想地需要應用,忽視任何比例性常數: Consider a set of stereo filters with left and right ear impulse responses h L (t) and h R (t), respectively. As described above, for monophonic mixing as described in equation (3) for ideal perfect tone compatibility, the following inequalities will ideally require application, ignoring any proportionality constants:

用於簡單之單音混合,理想上h L (t)+h R (t)=δ(t) (5) For simple mono mixing, ideally h L ( t ) + h R ( t ) = δ ( t ) (5)

我們稱為該立體聲輸出之單音混合當呈現時聽起來像該音頻輸入u(t)“單音播放相容性”或僅只單音相容性之 特性。除了單音播放相容性以外,其想要的是該hL(t)及hR(t)提供良好之立體聲化,亦即,該等輸出經由耳機之呈現聽起來自然,好像該聲音係來自該(等)虛擬喇叭位置及於一真實之收聽室中。其進一步想要的是容納該立體聲化音頻包括與不同虛擬喇叭位置及如此不同立體聲濾波器對一起混合之數個不同音頻輸入來源的案例。其將為想要的是該等單音濾波器係簡單以實現,且較佳地是與用於立體聲內容之單音下混合之一般實例相容。在該立體聲脈衝響應之方向及距離特性上沒有一顯著之衝擊,方程式(5)之限制大致上係不可能的。其暗指異於該濾波器脈衝響應之最初脈衝或分接,hR(t)=-hL(t),對於t>0。換句話說,當該立體聲濾波器被表達為具有脈衝響應hS(t)及hD(t)之和及差濾波器時,hS(t)=0,對於t>0。 The tone mix we call this stereo output sounds like the audio input u(t) "monophonic playback compatibility" or only the tone compatibility. In addition to monophonic playback compatibility, it is desirable that the h L (t) and h R (t) provide good stereoization, that is, the output sounds natural through the presentation of the headphones, as if the sound system From the (etc.) virtual horn position and in a real listening room. It is further desirable to accommodate the case where the stereo audio includes several different audio input sources mixed with different virtual horn positions and such different stereo filter pairs. It would be desirable for the monophonic filters to be simple to implement, and preferably compatible with the general example of subtone mixing for stereo content. There is no significant impact on the direction and distance characteristics of the stereo impulse response, and the limitation of equation (5) is generally not possible. It implies an initial pulse or tap that is different from the impulse response of the filter, h R (t) = -h L (t), for t > 0. In other words, when the stereo filter is expressed as having the sum of the impulse responses h S (t) and h D (t) and the difference filter, h S (t) = 0, for t > 0.

未馬上變得明顯的是此限制能以任何方式實現,而在該立體聲響應上沒有一顯著之衝擊。其需要該大部份之立體聲脈衝響應具有-1之相關係數。亦即,該脈衝響應將為與一變號完全相同的。 What is not immediately apparent is that this limitation can be achieved in any way without a significant impact on the stereo response. It requires that most of the stereo impulse response have a correlation coefficient of -1. That is, the impulse response will be exactly the same as a sign.

圖5以簡化形式顯示一典型之立體聲濾波器脈衝響應,大致用於該和濾波器hS(t)或用於該左或右耳立體聲濾波器。此一音響的脈衝響應之一般形式包括該直傳聲、一些早期反射、及緊接隔開的反射所組成之響應的一稍後部份,且如此被一擴散之回響很好地近似。 Figure 5 shows in a simplified form a typical stereo filter impulse response, generally for the sum filter h S (t) or for the left or right ear stereo filter. The general form of the impulse response of the sound includes a later portion of the response of the direct sound, some early reflections, and the closely spaced reflections, and is thus well approximated by a diffuse reverberation.

假設其設有分別具有脈衝響應hL0(t)及hR0(t)之左及右耳立體聲濾波器,且假設這些提供令人滿意之立體聲化 。本發明的一態樣係一組藉由脈衝響應hL(t)及hR(t)所界定之立體聲濾波器,其亦提供令人滿意之立體聲化、例如類似於一組已知之濾波器hL0(t)及hR0(t),但當下混合至一單音信號時,其輸出聽起來亦不錯。所討論者係hL(t)及hR(t)如何與hL0(t)及hR0(t)作比較,及吾人將已知hL0(t)及hR0(t)而如何設計hL(t)及hR(t)。 It is assumed that it is provided with left and right ear stereo filters having impulse responses h L0 (t) and h R0 (t), respectively, and it is assumed that these provide satisfactory stereoization. One aspect of the present invention is a set of stereo filters defined by impulse responses h L (t) and h R (t) that also provide satisfactory stereo, such as a set of known filters. h L0 (t) and h R0 (t), but when mixed to a single tone signal, the output sounds good. How the discussion is how h L (t) and h R (t) compare with h L0 (t) and h R0 (t), and how we will know h L0 (t) and h R0 (t) h L (t) and h R (t).

該直接響應部份Direct response part

於左耳及右耳立體聲脈衝響應之每一個中,該直接之響應將該位準及時間差編碼至該二個別之耳朵,其主要係負責用於賦予該收聽者方向感覺。本發明家發現該等立體聲濾波器之直接頭部相關轉移函數(HRTF)部份的頻譜效應係不太嚴重。再者,一典型之HRTF亦包括一時延分量。其意指當該等立體聲化輸出被混合至一單音信號時,用於該單音信號之同等濾波器將不是最小相位,且將導入一些額外之頻譜修飾。本發明家發現這些延遲係相當短的,例如<1毫秒。如此,當立體聲化信號之輸出被混合至一單音信號時,雖然該等延遲確實產生一些頻譜修飾,本發明家發現此頻譜修飾大致上不太嚴重,且藉由該延遲所產生之任何離散的回音係相當不能感知的。因此,於本發明之一些具體實施例中,hL(t)及hR(t)之立體聲濾波器脈衝響應的直接部份--那些藉由該HRTFs所界定者--係與用於任何立體聲濾波器脈衝響應者、例如濾波器hL0(t)及hR0(t)相同。亦即,根據本發明的一些態樣所注意之立體 聲濾波器hL(t)及hR(t)之特性排除該等立體聲濾波器之脈衝響應的直接部份。 In each of the left and right ear stereo impulse responses, the direct response encodes the level and time difference to the two individual ears, which are primarily responsible for imparting a sense of direction to the listener. The inventors have found that the spectral effects of the direct head related transfer function (HRTF) portion of the stereo filters are less severe. Furthermore, a typical HRTF also includes a delay component. It means that when the stereo output is mixed to a tone signal, the equivalent filter for the tone signal will not be the minimum phase and some additional spectral modifications will be introduced. The inventors have found that these delays are quite short, such as < 1 millisecond. Thus, when the output of the stereo signal is mixed to a tone signal, although the delay does produce some spectral modifications, the inventors have found that the spectral modification is substantially less severe and any dispersion resulting from the delay The echo system is quite unperceivable. Thus, in some embodiments of the invention, the direct portions of the stereo filter impulse responses of h L (t) and h R (t) - those defined by the HRTFs - are used for any The stereo filter impulse responders, such as filters h L0 (t) and h R0 (t), are identical. That is, the characteristics of the stereo filters h L (t) and h R (t) noted in accordance with some aspects of the present invention exclude the direct portion of the impulse response of the stereo filters.

注意於一些另外之具體實施例中,考慮此頻譜修飾。在橫越該虛擬喇叭位置給與一激發之左及右耳的結果,藉由考慮該組合頻譜,一具體實施例包括一補償等化濾波器,以達成一較平坦之頻譜響應。這通常被稱為補償該擴散之場域頭部響應,且如何承載此濾波對於那些熟諳該技藝者將為易懂的。雖然此補償能移去部份該頻譜立體聲暗號,其確實導致頻譜著色。 Note that in some other specific embodiments, this spectral modification is considered. The result of giving an excited left and right ear across the virtual horn position, by considering the combined spectrum, a particular embodiment includes a compensation equalization filter to achieve a flatter spectral response. This is often referred to as compensating for the field response of the spread, and how this filter is carried will be readily apparent to those skilled in the art. Although this compensation removes some of the spectral stereo ciphers, it does cause spectral coloring.

於一具體實施例中,該直傳聲響應係用於t<0者。亦即,h L (t)=h L0(t),用於t<3毫秒,及 (10) h R (t)=h R0(t),用於t<3毫秒。 (11) In a specific embodiment, the direct sound response is used for t<0. That is, h L ( t )= h L 0 ( t ) for t<3 milliseconds, and (10) h R ( t )= h R 0 ( t ) for t<3 milliseconds. (11)

現在考慮分別標示為hS0(t)及hDO(t)之原始和及差濾波器,與分別標示為hS(t)及hD(t)之立體聲化器的和及差濾波器。方程式(8a)及(9a)與圖4B敘述該左耳及右耳立體聲化器脈衝響應與該和及差濾波器脈衝響應間之向前及逆反關係,換句話說,其一脈衝響應係另一脈衝響應之混洗版本。又注意於一混洗操作及逆反混洗操作的實用之實施中,其於每一操作中不能包括該因數,但當作一範例,僅只決定一混洗中之和及差,且在該混洗中逆反該操作,被除以二,如於方程式(8b)及(9b)中所敘述。 Now consider the sum and difference filters labeled h S0 (t) and h DO (t), respectively, and the sum and difference filters of the stereos denoted as h S (t) and h D (t), respectively. Equations (8a) and (9a) and FIG. 4B describe the forward and reverse relationship between the left ear and right ear stereolizer impulse response and the sum and difference filter impulse response. In other words, one impulse response is another A shuffled version of an impulse response. Also paying attention to the practical implementation of a shuffling operation and a reverse shuffling operation, which cannot be included in each operation. The factor, but as an example, only determines the sum and difference in a shuff, and reverses the operation in the shuffling, divided by two, as described in equations (8b) and (9b).

本發明家發現該等典型之立體聲濾波器脈衝響應於該和及差濾波器中具有一類似信號能量。於方程式(5)中所 認知之單音相容性限制係等同於陳述該和濾波器沒有脈衝響應,亦即,hS(t)=0,用於t>0。用於不考慮該響應之該直接部份恆定的具體實施例,該需求係減少至如方程式(10)及(11)所顯示,即hS(t)=0,用於t>3毫秒或甚至稍後。 The inventors have found that these typical stereo filter impulses have a similar signal energy in response to the sum and difference filters. The tone compatibility constraint as recognized in equation (5) is equivalent to stating that the filter has no impulse response, that is, h S (t) = 0 for t > 0. For a specific embodiment that does not consider the direct partial constant of the response, the demand is reduced to as shown by equations (10) and (11), ie h S (t) = 0, for t > 3 milliseconds or Even later.

為了在該等和及差濾波器中維持大約相同之能量,與該原始濾波器作比較,該差頻道應被提升達大約3分貝,如果針對該等被修改響應中之反射能量必需維持該正確之頻譜及比率。然而,此修改造成該立體聲成像的一不想要之降級。該耳間的交互關係中之激變具有一強烈之感覺效果,且摧毀大部份空間及距離之感覺。 In order to maintain approximately the same energy in the sum and difference filters, the difference channel should be boosted by approximately 3 decibels as compared to the original filter, if the reflected energy in the modified response must be maintained correctly Spectrum and ratio. However, this modification caused an undesirable degradation of the stereo imaging. The catastrophe in the interaction between the ears has a strong sensory effect and destroys most of the space and distance.

於一具體實施例中,h D (t)=h D0(t),用於t之小值,比如說t<3毫秒,及 (12) ,用於t之大值,例如t>40毫秒。 (13) In a specific embodiment, h D ( t )= h D 0 ( t ) is used for a small value of t, such as t<3 milliseconds, and (12) , for a large value of t, such as t> 40 milliseconds. (13)

該等立體聲濾波器具有一差濾波器脈衝響應,其係一典型用於該脈衝響應之直接部份的立體聲差濾波器脈衝響應之3分貝提升,例如<3毫秒;及於該差濾波器脈衝響應之反射部份的稍後部份中具有一平坦之恆定值脈衝響應。 The stereo filters have a difference filter impulse response that is a 3 dB boost of the stereo differential filter impulse response typically used for the direct portion of the impulse response, such as <3 milliseconds; and the differential filter impulse response A flat constant value impulse response is present in a later portion of the reflected portion.

本發明家發現由hD(t)=hD0(t)變化至突然地發生,與該等原始濾波器作比較,該等結果之立體聲濾波器具有該立體聲成像的一不想要之降級。該耳間的交互關係中之激變具有一強烈之感覺效果,且摧毀大部份空間及距離之感覺。 The inventors found that h D (t) = h D0 (t) changes to Suddenly, compared to the original filters, the resulting stereo filters have an undesirable degradation of the stereo imaging. The catastrophe in the interaction between the ears has a strong sensory effect and destroys most of the space and distance.

此揭示內容的一態樣係以感知被遮蔽之逐漸方式於該立體聲響應之稍後部份中導入單音相容性限制,且如此在該立體聲成像上具有最小之衝擊。 One aspect of this disclosure is to introduce a tone compatibility limit in a later portion of the stereo response in a gradual manner in which the perception is masked, and thus has minimal impact on the stereo image.

本發明家發現立體聲濾波器對之典型立體聲房間脈衝響應最初典型係清楚有相互關係的,且於該響應之稍後部份中變得無關聯。再者,由於該較短之波長,該響應之較高頻率部份稍早於該立體聲響應中變得無關聯的。亦即,本發明家發現有一時間相依現象。 The inventors have found that the typical stereo room impulse response of a stereo filter is initially well correlated and becomes uncorrelated in later portions of the response. Moreover, due to the shorter wavelength, the higher frequency portion of the response becomes uncorrelated earlier than the stereo response. That is, the inventors found a time-dependent phenomenon.

於本發明的一具體實施例中,該立體聲對之和濾波器係藉由一時變濾波器與一典型立體聲濾波器對之典型和濾波器有關。藉由f(t,τ)標示該時變濾波器之時變脈衝響應,其係在時間t對一在時間t=τ之脈衝、亦即對輸入δ(t-τ)的時變濾波器之響應。亦即,h S (t)=ʃh S0(t-τ)f(t,τ). (14)在此f(t,τ)係使得f(0,τ)=δ(τ)及 (15) ,用於稍後時間,例如t>40毫秒,或t>80毫秒 (16) In a specific embodiment of the invention, the stereo pair filter is related to a typical filter of a typical stereo filter pair by a time varying filter. The time-varying impulse response of the time-varying filter is denoted by f(t, τ), which is a time-varying filter at time t=one pulse at time t=τ, ie, input δ(t-τ) The response. That is, h S ( t )=ʃ h S 0 ( t - τ ) f ( t , τ ). (14) where f(t, τ) is such that f (0, τ ) = δ ( τ ) and (15) For later time, such as t>40 milliseconds, or t>80 milliseconds (16)

於一些具體實施例中,F(t,τ)為或近似零延遲、線性相位、低通濾波器脈衝響應,具有標示為Ω(t)>0之減少的時間相依頻寬,使得標示為|F(t,ω)|之時間相依頻率響應對於該頻寬以下之低頻具有該|F(t,ω)|係平坦之特性,且在該頻寬之外為0。 In some embodiments, F(t,τ) is or approximates a zero-delay, linear phase, low-pass filter impulse response having a time-dependent bandwidth labeled Ω(t)>0, such that the flag is | The time-dependent frequency response of F(t, ω)| has a characteristic that the |F(t, ω)| is flat for a low frequency below the bandwidth, and is 0 outside the bandwidth.

,用於|ω|<Ω(t)| (17) For | ω |<Ω( t )| (17)

,用於|ω|>Ω(t) (18) For | ω |> Ω( t ) (18)

在此該時變頻率響應被標示為F(t,ω),具有 Here the time varying frequency response is labeled F(t, ω) with

且在此該時變頻寬係在時間中單調遞減,亦即,Ω(t 1)>Ω(t 2),用於t 1<t 2 (20) At this time, the variable frequency width is monotonically decreasing in time, that is, Ω( t 1 )>Ω( t 2 ) for t 1 < t 2 (20)

一具體實施例使用一濾波器時間相依頻寬由在t=0之至少20kHz單調地增加至用於高時間值、例如用於t>10毫秒的大約100Hz或更少。亦即,使得,用於t>40毫秒 (21) A particular embodiment uses a filter time dependent bandwidth to monotonically increase from at least 20 kHz at t = 0 to about 100 Hz or less for high time values, such as for t > 10 milliseconds. That is, and For t>40 ms (21)

那些熟諳該技藝者將再次了解被表達於方程式(14)-(21)中之濾波器的形式係於連續之時間中。在離散之時間項中敘述此將為相當易懂的,故將不在此中被討論,以便不會由敘述本發明之特色分散。 Those skilled in the art will again understand that the forms of the filters expressed in equations (14)-(21) are in continuous time. It will be fairly straightforward to describe this in discrete time terms and will not be discussed herein so as not to be disambiguated by the nature of the invention.

關於該差濾波器,一具體實施例使用一差濾波器,其脈衝響應hD(t)係與一差濾波器有關,該差濾波器之空間化將被匹配為 在此hD0(t)標示該原始差濾波器脈衝響應。 With respect to the difference filter, a particular embodiment uses a difference filter whose impulse response h D (t) is related to a difference filter whose spatialization will be matched to Here h D0 (t) indicated that the original difference filter impulse response.

那些熟諳該技藝者將再次了解被表達於方程式(22)中之濾波器的形式係於連續之時間中。在離散之時間項中敘述此將為相當易懂的,故將不在此中被討論,以便不會由敘述本發明之特色分散。 Those skilled in the art will again understand that the form of the filter expressed in equation (22) is in continuous time. It will be fairly straightforward to describe this in discrete time terms and will not be discussed herein so as not to be disambiguated by the nature of the invention.

具有方程式(22)之脈衝響應的濾波器係適當的,在此標示為f(t,τ)之低通濾波器脈衝響應具有零延遲及線性相位,以致其空間化品質待匹配之原始差濾波器hD0(t)及該差濾波器hD(t)係相位同調的。 The filter having the impulse response of equation (22) is suitable, and the low-pass filter impulse response denoted here as f(t, τ) has zero delay and linear phase, so that its spatial quality is matched to the original difference filter. The device h D0 (t) and the difference filter h D (t) are phase-coherent.

注意因為f(0,τ)=δ(τ),h D (0)=h D0(0) Note that since f (0, τ ) = δ ( τ ), h D (0) = h D 0 (0)

再者,因為f(t,τ)0,用於稍後時間,例如t>40毫秒,,用於t>40毫秒左右。 Again, because f(t, τ) 0, for later time, for example t>40 milliseconds, For t>40ms or so.

因此,在晚些時候,例如在40毫秒,該差濾波器脈衝響應係與該待匹配或典型立體聲濾波器之差濾波器成比例。如此,對該原始差濾波器脈衝響應hD0(t)之修改在該差頻道上實現一頻率相依提升,該差頻道在0分貝於界定為t=0之最初脈衝時間開始,且在漸進地較低頻率於時間t增加時增加至+3分貝。在該和及差濾波器將具有於振幅中類似及無關聯之脈衝響應的假設之下,此增益係適當的。雖然這未總是唯一的,本發明家已發現這為一合理之假設,且已發現該差異頻道脈衝響應hD(t)及一對立體聲濾波器的差頻道脈衝響應間之關係,其空間化將被匹配一合理之方式,以修正該頻譜及針對該等被修改濾波器之混響 比。 Thus, at a later time, for example at 40 milliseconds, the difference filter impulse response is proportional to the difference filter of the to-be-matched or typical stereo filter. Thus, the modification of the original difference filter impulse response h D0 (t) achieves a frequency dependent boost on the difference channel, the difference channel starting at 0 dB at the initial pulse time defined as t=0, and progressively The lower frequency increases to +3 decibels as time t increases. This gain is appropriate under the assumption that the sum and difference filters will have similar and uncorrelated impulse responses in amplitude. Although this is not always unique, the inventors have found this to be a reasonable assumption and have found the relationship between the differential channel impulse response h D (t) and the difference channel impulse response of a pair of stereo filters. The ration will be matched in a reasonable manner to correct the spectrum and the reverberation ratio for the modified filters.

然而,本發明不被限制於方程式(14)及(22)中所顯示之關係。於另一具體實施例中,其他關係可被使用於進一步改善與任何所提供或決定之立體聲濾波器對、例如與脈衝響應hLO(t)及hRO(t)的頻譜匹配。此特定之方式在此中被呈現為一相當簡單之方法,以達成一合理之結果,且不意指為其限制。 However, the present invention is not limited to the relationship shown in equations (14) and (22). In another embodiment, other relationships may be used to further improve spectral matching with any provided or determined stereo filter pair, such as with impulse responses h LO (t) and h RO (t). This particular manner is presented herein as a relatively simple method to achieve a reasonable result and is not intended to be limiting.

該等目標立體聲濾波器能接著使用方程式(8a)與(9a)及圖4B或方程式(8b)與(9b)之混洗關係被重建。此方式已被發現,以於該單音混合中之回響減少、及該立體聲響應上之感覺遮蔽之間提供一有效的平衡。變遷至-1之相關係數平順地發生,且在一最初之時間間隔期間、例如該等脈衝響應之最初40毫秒。於此一具體實施例中,該單音混合中之回響響應被限制於約40毫秒,使該高頻回響係遠較短。 The target stereo filters can then be reconstructed using the shuffle relationship of equations (8a) and (9a) and FIG. 4B or equations (8b) and (9b). This approach has been found to provide an effective balance between the reduced reverberation in the tone mix and the perceived mask on the stereo response. The correlation coefficient that transitions to -1 occurs smoothly and during an initial time interval, such as the first 40 milliseconds of the impulse response. In this embodiment, the echo response in the tone mix is limited to about 40 milliseconds, making the high frequency reverberation system much shorter.

該40時間毫秒被建議用於該單音混合,而感覺幾乎無回聲的。雖然一些早期反射及回響可仍然存在於該單音混合中,這是藉由該直傳聲被有效地遮蔽,且本發明家已發現其不被感知為一離散之回音或額外之回響。 The 40 time millisecond is suggested for this tone mix, and feels almost echo free. Although some early reflections and reverberations may still be present in the monophonic mixture, this is effectively masked by the direct sound, and the inventors have discovered that it is not perceived as a discrete echo or additional reverberation.

本發明不被限制於該長度40毫秒之變遷區域。此變遷區域可視該應用而定被變更。如果其係想要模擬一具有特別長之回響時間、或低直傳聲對回響之比率的房間,該變遷時間可被進一步延長,且與用於此一房間之標準立體聲濾波器作比較,仍然對該單音相容性提供一改良。該 40變遷毫秒時間被發現適合用於一特定之應用,在此該等原始立體聲濾波器具有150毫秒之回響時間,且該單音混合係需要為盡可能接近無回聲。 The invention is not limited to this transition region of length 40 milliseconds. This transition area can be changed depending on the application. If the system wants to simulate a room with a particularly long reverberation time, or a low direct sound to reverberation ratio, the transition time can be further extended and compared to the standard stereo filter used in this room, still An improvement is provided to the tone compatibility. The The 40 transition millisecond time was found to be suitable for a particular application where the original stereo filters have an echo time of 150 milliseconds and the tone mixing system needs to be as close as possible to no echo.

雖然於一些具體實施例中,該和濾波器完全被消除,這不是一項要求。該和脈衝響應之振幅被一因數所減少,而足夠於該單音混合之回響部份中達成一顯著之差異或減少。本發明家將用於約6分貝的回響位準中之改變的“恰好值得注意的差異”選擇作為一準則。如此,於本發明的一些具體實施例中,與以用典型立體聲濾波器立體聲化信號的單音混合所發生者作比較,至少6分貝的和濾波器回響響應中之減少被使用。如此,於一些具體實施例中,該和濾波器不被完全地消除,但其之影響、例如其脈衝響應之振幅係顯著地減少,例如藉由使該和頻道濾波脈衝響應振幅衰減達6分貝或更多。藉由結合該原始和濾波器脈衝響應及該上面提出之被修改濾波器脈衝響應,以決定一標示為hS"(t)之和脈衝響應,一具體實施例達成此: Although in some embodiments the filter is completely eliminated, this is not a requirement. The amplitude of the sum pulse response is reduced by a factor sufficient to achieve a significant difference or reduction in the reverberant portion of the tone mix. The inventors have chosen the "just noticeable difference" for the change in the reverberation level of about 6 decibels as a criterion. Thus, in some embodiments of the invention, a reduction of at least 6 decibels and a filter echo response is used in comparison to the occurrence of a tone mixing with a stereo signal of a typical stereo filter. Thus, in some embodiments, the sum filter is not completely eliminated, but its effect, such as the amplitude of its impulse response, is significantly reduced, such as by attenuating the sum of the channel filter impulse response amplitude by up to 6 decibels. Or more. By combining the original and filter impulse responses and the modified filter impulse response set forth above to determine an impulse response labeled h s "(t), a specific embodiment achieves this:

用於β的一典型值係1/2,其同樣地加權該原始及被修改之和濾波器脈衝響應。於另一具體實施例中,其他加權被使用。 A typical value for β is 1/2, which equally weights the original and modified sum filter impulse response. In another embodiment, other weightings are used.

亦應注意的是f(t,τ)之限制為零延遲,且線性相位係用於方程式(22)之差頻道的混洗轉換及修改中之單純及適當的相位重建。對於熟練信號處理者應為明顯的是此限制可被放鬆,所提供之適當濾波亦施加至該差頻道,以於 hD(t)及hD0(t)之間建立一關係。藉由本發明家所作的一項觀察係該等正確之相位關係及立體聲響應的稍後部份中之方向暗號對於空間及距離的一般感覺為不重要的。因此,此濾波非為絕對地需要。如果該目標係維持該等立體聲濾波器hL(t)、hR(t)中之回響比率,如存在於另一對立體聲濾波器hL0(t)、hR0(t)中,則這可-於一頻率相依具體實施例中-藉由對該差濾波器脈衝響應hD(t)之一適當增益所達成。 It should also be noted that the limit of f(t, τ) is zero delay, and the linear phase is used for the simple and appropriate phase reconstruction in the shuffling conversion and modification of the difference channel of equation (22). It should be apparent to the skilled signal processor that this limitation can be relaxed and that the appropriate filtering provided is also applied to the difference channel to establish a relationship between h D (t) and h D0 (t). An observation made by the inventor is that the correct phase relationship and the direction sign in the later part of the stereo response are not important to the general perception of space and distance. Therefore, this filtering is not absolutely necessary. If the target maintains the reverberation ratio in the stereo filters h L (t), h R (t), if present in another pair of stereo filters h L0 (t), h R0 (t), then It may be in a frequency dependent embodiment - by an appropriate gain of one of the difference filter impulse responses h D (t).

圖6顯示一信號處理設備之簡化方塊圖,且圖7顯示操作一信號處理設備之方法的簡化流程圖。該設備將決定一組左耳信號hL(t)及右耳信號hR(t),該等信號形成一對立體聲濾波器之左耳及右耳脈衝響應,該立體聲濾波器近似具有左耳及右耳脈衝響應hL0(t)與hR0(t)的該對立體聲濾波器之立體聲化。該方法包括在703中接收一左耳信號hL0(t)及右耳信號hR0(t),該等信號代表對應左耳及右耳立體聲濾波器之脈衝響應,該等濾波器被組構成立體聲化一音頻信號,且其立體聲響應將被匹配。該方法另在705中包括混洗該左耳信號及右耳信號,以形成一與該左及右耳信號之和成比例的和信號、及一與該左耳信號和該右耳信號所之差成比例的差信號。於圖6之設備中,這是藉由混洗器603所進行。該方法在707中另包括藉由一時變濾波器(和濾波器)605過濾該和信號,該過濾形成一被過濾之和信號,且藉由一差時變濾波器607處理該差信號--差濾波器--其係以該和濾波器605為其特徵,該處理形成一 被過濾之差信號。該方法在709中另包括解混洗該被過濾之和信號及該被過濾之差信號,以形成與產生一分別與立體聲濾波器之左及右耳脈衝響應成比例的左耳信號及右耳信號,該等立體聲濾波器之空間化特徵匹配該等待匹配立體聲濾波器之之空間化特徵,且其輸出可與可接收之聲音被下混合至一單音混合。於圖6中,該解混洗器609係與具有達2之增加分割的混洗器603相同。該結果之脈衝響應界定被組構成立體聲化一音頻信號之立體聲濾波器,且進一步具有該和頻道脈衝響應平順地減少至一不能感知之位準的特性,例如在該第一40毫秒左右超過-6分貝,且該差頻道變遷至變得於該第一40毫秒左右與一典型或特別之待匹配立體聲濾波器差頻道脈衝響應成比例。 Figure 6 shows a simplified block diagram of a signal processing device, and Figure 7 shows a simplified flow chart of a method of operating a signal processing device. The device will determine a set of left ear signals h L (t) and right ear signals h R (t) that form a left ear and right ear impulse response of a pair of stereo filters having approximately a left ear And the right ear impulse response h L0 (t) and h R0 (t) stereo pairing of the stereo filter. The method includes receiving a left ear signal h L0 (t) and a right ear signal h R0 (t) at 703, the signals representing impulse responses of corresponding left and right ear stereo filters, the filters being grouped Stereo an audio signal and its stereo response will be matched. The method further includes shuffling the left ear signal and the right ear signal to form a sum signal proportional to the sum of the left and right ear signals, and a left ear signal and the right ear signal. A poorly proportional difference signal. In the apparatus of Figure 6, this is done by the shuffler 603. The method further includes, in 707, filtering the sum signal by a time varying filter (and filter) 605, the filtering forming a filtered sum signal, and processing the difference signal by a differential time varying filter 607- The difference filter is characterized by the sum filter 605 which forms a filtered difference signal. The method further includes, in 709, deshuffling the filtered sum signal and the filtered difference signal to form a left ear signal and a right ear that are proportional to generating a left and right ear impulse response respectively to the stereo filter. Signals, the spatialized features of the stereo filters match the spatialized features of the wait-matched stereo filter, and the output can be downmixed with a receivable sound to a single tone mix. In Figure 6, the de-mixer 609 is the same as the shuffler 603 having an increased split of up to two. The resulting impulse response defines a stereo filter that is grouped to form a stereo-audio signal, and further has the characteristic that the sum-channel impulse response is smoothly reduced to an insensible level, such as over the first 40 milliseconds or more - 6 decibels, and the difference channel transitions to become approximately proportional to a typical or special stereo signal difference channel impulse response to be matched for the first 40 milliseconds.

如此已敘述一操作信號處理設備之方法。該方法包括接收代表被組構成立體聲化一音頻信號的對應之立體聲濾波器對的脈衝響應之一對信號。該方法包括藉由一對濾波器處理該對接收信號,該對濾波器之每一個以一具有時變濾波器特徵之修改濾波器為其特徵,該處理形成代表一對應對被修改之立體聲濾波器的脈衝響應之一對被修改的信號。該等被修改之立體聲濾波器被組構成立體聲化一音頻信號,且進一步於該單音混合中具有一低感知回響、及透過耳機在該立體聲濾波器上之最小衝擊的特性。 A method of operating a signal processing device has thus been described. The method includes receiving a pair of signals representative of an impulse response of a pair of corresponding stereo filters that are grouped to form a stereo-audio signal. The method includes processing the pair of received signals by a pair of filters each characterized by a modified filter having a time varying filter characteristic that forms a pair of stereo filters that are modified to be modified One of the impulse responses of the device is the modified signal. The modified stereo filters are grouped to form a stereo-audio signal, and further have a low perceived reverberation in the mono mix and a minimum impact characteristic of the earphone on the stereo filter.

根據本發明之一或更多態樣的立體聲濾波器具有該等性質: A stereo filter according to one or more aspects of the present invention has these properties:

●該等脈衝響應之直接部份、例如於該脈衝響應之最初 3至5毫秒中,係藉由該等虛擬喇叭位置之頭部相關轉移函數所界定。 a direct portion of the impulse response, for example at the beginning of the impulse response In 3 to 5 milliseconds, it is defined by the head related transfer function of the virtual horn positions.

●與該差濾波器脈衝響應作比較,在該和濾波脈衝響應中顯著地減少之層次及/或顯著地較短之回響時間。 • Comparing the differential filter impulse response with a significantly reduced level and/or a significantly shorter reverberation time in the summed filter impulse response.

●由該和濾波器之脈衝響應的直接部份平順變遷至該和濾波器的稍後零或可忽略之響應部份。該平順之變遷係隨著時間之消逝選擇的頻率。 • The direct portion of the impulse response of the sum filter transitions smoothly to a later zero or negligible response portion of the sum filter. The smooth transition is the frequency of selection as time passes.

這些性質將不會發生在任何實用之房間響應中,且如此將不會被呈現於典型或待匹配立體聲濾波器中。這些性質被導入、或設計成一組立體聲濾波器。 These properties will not occur in any practical room response and will not be presented in a typical or to-be-matched stereo filter. These properties are imported or designed into a set of stereo filters.

這些性質係在下面更詳細地敘述。 These properties are described in more detail below.

喇叭相容性Speaker compatibility

雖然上面之敘述描述具有單音播放相容性之立體聲濾波器,本發明之另一態樣係具有根據本發明之一具體實施例的濾波器之輸出信號立體聲化器係亦與透過一組揚聲器之播放相容。 Although the above description describes a stereo filter having monophonic playback compatibility, another aspect of the present invention has an output signal stereoizer of a filter according to an embodiment of the present invention and also transmits a set of speakers. Play compatible.

音響的串音係用於敘述當收聽一對立體揚聲器時的現象之術語,例如在大約一收聽者之中心前面,該收聽者之每一耳朵將由該等立體揚聲器之兩者接收信號。具有根據本發明之具體實施例的立體聲濾波器,該音響的串音造成該較低頻率回響之一些消去。大致上,一回響的響應對一輸入之稍後部份變得漸進地低通過濾的。如此,當透過喇叭試聽時,以根據本發明之具體實施例的立體聲濾波器濾 波之立體聲化信號已被發現聽起來較少回響。這特別是相當小地緊接隔開立體喇叭之案例,諸如可在一行動媒體裝置中發現者。 The crosstalk of the sound is used to describe the terminology of a phenomenon when listening to a pair of stereo speakers, for example, in front of the center of a listener, each ear of the listener will receive signals from both of the stereo speakers. There is a stereo filter in accordance with a particular embodiment of the present invention, the crosstalk of the acoustics causing some cancellation of the lower frequency reverberation. In general, a reverberant response becomes progressively lower through the filtering of a later portion of an input. Thus, when listening through a horn, a stereo filter is used in accordance with a specific embodiment of the present invention. The stereo signal of the wave has been found to sound less reverberant. This is particularly the case in which the stereo speakers are closely spaced, such as can be found in a mobile media device.

複雜性減少Reduced complexity

其係已知設計涉及相當少之計算的立體聲濾波器,以藉由使用一脈衝響應之回響部份係對於空間位置較不敏感的觀察來實施。如此,很多立體聲處理系統使用其脈衝響應具有一共同之結尾部份的立體聲濾波器,而用於該等不同模擬之虛擬喇叭位置。譬如看前述世界專利公告第WO 9914983及WO 9949574號。本發明之具體實施例係可適用於此等立體聲處理系統,且修改此等立體聲濾波器,以具有單音播放相容性。特別地是,根據本發明的一些具體實施例所設計之立體聲濾波器具有該左及右耳脈衝響應之回響結尾的稍後部份係不同相之特性,數學表示為hR(t)-hL(t),用於時間t>40毫秒左右。因此,根據該等立體聲濾波器之一相當低計算複雜性實施,僅只單一濾波脈衝響應需要被決定用於該響應之稍後部份,且用於所有虛擬喇叭位置,此被決定稍後部份脈衝響應係可用於立體聲濾波器對之左及右耳脈衝響應的每一個中,導致記憶體及計算中之節省。每一個此立體聲濾波器對之和濾波器包括一逐漸之時變截止頻率,其進一步延伸該和濾波器低頻內容進入該立體聲響應。 It is known that the design involves a relatively small number of computational stereo filters to be implemented by using an echo response portion of the impulse response that is less sensitive to spatial position. As such, many stereo processing systems use stereo filters whose impulse responses have a common end portion for the virtual horn positions of the different analogs. See, for example, the aforementioned World Patent Publication No. WO 9914983 and WO 9949574. Embodiments of the present invention are applicable to such stereo processing systems and modify such stereo filters to have monophonic playback compatibility. In particular, the stereo filter designed in accordance with some embodiments of the present invention has the characteristics of the different phases of the reverberation end of the left and right ear impulse responses, mathematically expressed as h R (t) -h L (t) for time t > 40 ms. Therefore, according to a relatively low computational complexity implementation of one of the stereo filters, only a single filtered impulse response needs to be determined for later portions of the response and used for all virtual horn positions, which is determined later. The impulse response can be used in each of the left and right ear impulse responses of the stereo filter pair, resulting in savings in memory and computation. Each of the stereo filter pair sum filters includes a gradual time varying cutoff frequency that further extends the summed filter low frequency content into the stereo response.

一示範演算法及結果An exemplary algorithm and results

該先前段落提出一般之性質及方式,以達成該被修改之立體聲濾波。雖然濾波器之設計及處理有很多可能之變化,其將具有類似結果,以下之示範例被呈現,以示範該想要之濾波器性質,且提供修改一組現存立體聲濾波器之較佳方式。 This previous paragraph presents general nature and manner to achieve this modified stereo filtering. While there are many possible variations in the design and processing of filters, which will have similar results, the following examples are presented to demonstrate the desired filter properties and provide a preferred way to modify a set of existing stereo filters.

圖8顯示MATLAB(麻薩諸塞州內迪克市之Mathworks公司)語法中之編碼的一部份,其實行將一對立體聲濾波器脈衝響應轉換成代表立體聲濾波器的脈衝響應之信號的方法。該線性相位、零延遲、時變低通濾波器係使用一系列序連的第一階濾波器實施。此簡單之方式近似一高斯濾波器。MATLAB編碼之此簡短段落採取一對立體聲濾波器h_LO及h_RO,且建立一組輸出立體聲濾波器h_L及h_R。其係基於48kHz之取樣比率。 Figure 8 shows a portion of the encoding in the MATLAB (Mathworks, Inc., Nadick, Mass.) grammar that implements a method of converting a pair of stereo filter impulse responses into signals representative of the impulse response of a stereo filter. The linear phase, zero delay, time varying low pass filter is implemented using a series of sequential first order filters. This simple approach approximates a Gaussian filter. This short paragraph of MATLAB encoding takes a pair of stereo filters h_LO and h_RO and creates a set of output stereo filters h_L and h_R. It is based on a sampling ratio of 48 kHz.

首先,於803中,該等輸入濾波器被混洗,以建立該原始之和及差濾波器。(看該編碼的1-2行) First, in 803, the input filters are shuffled to establish the original sum and difference filters. (Look at the 1-2 lines of the code)

該高斯濾波器(B)之3分貝頻寬係隨著該樣本數目之平方反比及適當的按比例增減係數而變化。由此計算該高斯濾波器之相關變異數(GaussVar),且除以四,以獲得該指數的第一階濾波器之變異數(ExponVar)。於805中,這被使用於計算該時變指數加權因數(a)。(看該編碼的3-6行) The 3 dB bandwidth of the Gaussian filter (B) varies with the square inverse of the number of samples and the appropriate scaling factor. The correlation variation (GaussVar) of the Gaussian filter is thus calculated and divided by four to obtain the first-order filter variation (ExponVar) of the index. In 805, this is used to calculate the time varying exponential weighting factor (a). (Look at lines 3-6 of the code)

該濾波器係在807中使用該第一階濾波器之二向前及二逆反通過實施。該和及差異響應兩者被濾波。(看該編 碼的7-12行) The filter is implemented in 807 using the first and second inverses of the first order filter. Both the sum and the difference response are filtered. (see the editor 7-12 lines of code)

於809中,該差異由該原始差響應之按比例上升版本重做,少於該被濾波之差響應的一適當數量。這實際上係由在時間零之0分貝至該稍後響應中的+3分貝之差頻道的一頻率選擇性提升。(看該編碼的13行) In 809, the difference is redone from the scaled up version of the original difference response, less than an appropriate amount of the filtered difference response. This is actually a frequency selective boost from the difference of 0 dB in time zero to +3 dB in the later response. (Look at the 13 lines of the code)

最後於811中,該等濾波器被重新混洗,以建立該被修改之左及右立體聲濾波器。(看該編碼的14-15行) Finally in 811, the filters are re-washed to create the modified left and right stereo filters. (Look at lines 14-15 of the code)

用於一定位在該收聽者之前面的聲音,以下之圖面係由圖8中之編碼方法對一組立體聲濾波器脈衝響之應用所獲得,而具有150毫秒之最大回響時間及約13分貝之直接對回響能量的比率。 For a sound positioned in front of the listener, the following picture is obtained by the encoding method of Figure 8 for a set of stereo filter pulse applications, with a maximum reverberation time of 150 milliseconds and approximately 13 decibels. The ratio of direct to reverberant energy.

圖9顯示該時變濾波器f(t,τ)之脈衝響應對一脈衝在數個時間τ的繪圖:τ在1、5、10、20及40毫秒。該首先二脈衝係超出該圖面之直立比例。圖9清楚地顯示所施加之濾波器脈衝響應的高斯近似值、及該大約高斯濾波器脈衝響應隨著時間增加之變異數。既然該第一階濾波器係向後與向前兩者延伸,該結果之濾波器近似一零延遲、線性相位、低通濾波器。 Figure 9 shows the plot of the impulse response of the time varying filter f(t, τ) versus a pulse over several times τ: τ at 1, 5, 10, 20 and 40 ms. The first two pulses are outside the erect ratio of the drawing. Figure 9 clearly shows the Gaussian approximation of the applied filter impulse response and the variance of the approximately Gaussian filter impulse response over time. Since the first order filter extends both backwards and forwards, the resulting filter approximates a zero delay, linear phase, low pass filter.

圖10顯示脈衝響應之時變濾波器f(t,τ)的頻率響應能量在1、5、10、20及40毫秒之時間τ的繪圖。於此大約由0至3毫秒之案例中,其能被看出該響應之直接部份將大部份未受該濾波器影響的,而達40毫秒時,該濾波器直至100Hz造成幾乎10分貝之衰減。因為該脈衝響應之大約高斯形狀,該頻率響應亦具有一大約之高斯輪廓。此 大約之高斯頻率響應輪廓、及該截止頻率隨著時間之消逝的變化兩者有助於達成對該原始濾波器所造成之修改的感覺遮蔽。 Figure 10 shows a plot of the frequency response energy of the impulse response time varying filter f(t, τ) at times τ of 1, 5, 10, 20 and 40 ms. In the case of about 0 to 3 milliseconds, it can be seen that the direct part of the response will be largely unaffected by the filter, and up to 40 milliseconds, the filter will cause almost 10 dB up to 100 Hz. Attenuation. Because of the approximately Gaussian shape of the impulse response, the frequency response also has an approximate Gaussian profile. this Approximately the Gaussian frequency response profile, and the change in the cutoff frequency over time, contributes to achieving a perceived occlusion of the modifications made to the original filter.

圖11顯示該原始的左耳脈衝響應hL0(t)及被修改之左耳脈衝響應hL(t)。其為明顯的是兩者具有一類似位準之回響能量。該直傳聲保持恆定的。注意該直傳聲之最初脈衝測量約0.2,且不能被顯示在該圖面中之比例上。 Figure 11 shows the original left ear impulse response h L0 (t) and the modified left ear impulse response h L (t). It is obvious that both have a similar level of reverberation energy. This direct transmission remains constant. Note that the initial pulse of the direct sound is measured to be about 0.2 and cannot be displayed on the scale in the drawing.

圖12顯示該原始及被修改之總和脈衝響應hS0(t)及hS(t)的比較。這清楚地示範該總和響應之減少位準及回響時間。當該輸出被混合成單音時,這是在該回響中達成一顯著減少之特色。其亦可被看出該被修改之總和響應hS(t)變成被漸進地之過濾低通,僅只具有延伸超出該響應之早期部份的最低頻率信號分量。 Figure 12 shows a comparison of the original and modified summed impulse responses h S0 (t) and h S (t). This clearly demonstrates the reduced level and reverberation time of the sum response. When the output is mixed into a single tone, this is a feature that achieves a significant reduction in the reverberation. It can also be seen that the modified sum response h S (t) becomes a progressively filtered low pass with only the lowest frequency signal component extending beyond the early portion of the response.

圖13顯示該原始及被修改之差脈衝響應hD0(t)及hD(t)。其能被觀察到該差信號係在位準中被提升。這是達成該二響應之可比較的頻譜。 Figure 13 shows the original and modified difference impulse responses h D0 (t) and h D (t). It can be observed that the difference signal is boosted in the level. This is a comparable spectrum that achieves the second response.

該等立體聲濾波器之時頻分析Time-frequency analysis of these stereo filters

根據本發明之一或更多態樣的立體聲濾波器、例如以一對立體聲脈衝響應為其特徵者,當使用於濾波一來源信號、例如藉由以該立體聲脈衝響應之卷積或以別的方式施加至一來源信號時,加入模擬至一經由耳機收聽之收聽者的方向、距離及房間音響學之空間品質。 A stereo filter according to one or more aspects of the present invention, such as characterized by a pair of stereo impulse responses, when used to filter a source signal, for example by convolution with the stereo impulse response or otherwise When the mode is applied to a source signal, the direction of the listener, the distance to the listener via headphones, and the spatial quality of the room sound are added.

在可重疊的區段信號上之時頻分析、例如使用該短時 傅立葉轉換或其他短時轉換於該技藝係熟知的。譬如,頻率-時間分析繪圖係已知為頻譜圖。一短時傅立葉轉換、例如於典型地遍及一段想要之信號實施為一窗口式離散傅立葉轉換(DFT)中。其他轉換亦可被使用於時頻分析,例如子波轉換及其他轉換。一脈衝響應係一時間信號,且因此能以其時頻性質為其特徵。本發明之立體聲濾波器可藉由此時頻特徵所敘述。 Time-frequency analysis over overlapping segment signals, for example using the short time Fourier transforms or other short-term transitions are well known in the art. For example, frequency-time analysis plots are known as spectrograms. A short time Fourier transform, for example, is typically implemented as a windowed discrete Fourier transform (DFT) throughout a desired signal. Other conversions can also be used for time-frequency analysis, such as wavelet conversion and other conversions. An impulse response is a time signal and can therefore be characterized by its time-frequency nature. The stereo filter of the present invention can be described by this time-frequency feature.

當下混合至單一輸出時,根據本發明之一或更多態樣的立體聲濾波器被組構成可透過耳機同時地達成一使人信服之立體聲效果,例如根據一對待匹配立體聲濾波器、及一單音播放相容之信號。本發明之立體聲濾波器具體實施例被組構成具有該特性,即該等立體聲濾波器脈衝響應之(短時)頻率響應隨著時間之消逝以一或多個特色而變化。特別地是,該和濾波器脈衝響應、例如該二左及右立體聲濾波器脈衝響應之算術和,隨著時間之消逝具有一圖案及頻率,該頻率與該等差濾波器脈衝響應、例如該左及右立體聲濾波器脈衝響應之算術差顯著地不同。用於一典型之立體聲響應,該等和及差濾波器於隨著時間之消逝的頻率響應中顯示一很類似之變化。該響應之早期部份包含大多數該能量,且該稍後響應包含該回響或擴散分量。其係該早期及稍後部份、與該等濾波器的有特色結構間之平衡,該結構賦予該脈衝響應之空間或立體聲特徵。然而,當混合至單音時,此回響響應通常使該信號清晰度及感知品質降級。 When downmixed to a single output, a stereo filter according to one or more aspects of the present invention is configured to simultaneously achieve a convincing stereo effect through the earphone, for example, according to a stereo signal to be matched, and a single The sound plays a compatible signal. The stereo filter embodiments of the present invention are grouped to have the characteristic that the (short-term) frequency response of the impulse response of the stereo filters varies over time with one or more characteristics. In particular, the sum of the sum of the filter impulse response, for example, the two left and right stereo filter impulse responses, has a pattern and frequency as time elapses, the frequency and the differential filter impulse response, for example, The arithmetic differences between the left and right stereo filter impulse responses are significantly different. Used for a typical stereo response, the sum and difference filters show a very similar change in the frequency response over time. The early portion of the response contains most of this energy, and the later response contains the reverberation or diffusion component. It is the balance between this early and later part, with the distinctive structure of the filters, which gives the spatial or stereo characteristics of the impulse response. However, when mixed to a single tone, this reverberation response typically degrades the signal clarity and perceived quality.

藉由簡單之相容性係意指保有該方程式(5)。亦即,異於用在該濾波器脈衝響應之最初脈衝或分接,hR(t)=-hL(t),用於t>0亦即,該hS(t)=0,用於t>0。該結果之濾波器組被稱為過分簡化之單音播放相容濾波器組、或過分簡化之濾波器。 By simple compatibility is meant to retain the equation (5). That is, different from the initial pulse or tap used in the impulse response of the filter, h R (t)=-h L (t), for t>0, ie, h S (t)=0, At t>0. The resulting filter bank is referred to as an oversimplified monophonic playback compatible filter bank, or an oversimplified filter.

於此段落中敘述本發明之立體聲濾波器對的此等脈衝響應之時頻分析的一些特徵,且提供用於一些時頻參數的一些典型值與諸值之範圍。這是被示範資料所示範及比較於:1)一組待匹配、例如典型之立體聲濾波器,及2)藉由強加簡單之相容性自該等典型立體聲濾波器所導出之一濾波器組,以獲得一過分簡化之單音相容性濾波器組。 Some features of the time-frequency analysis of these impulse responses of the pair of stereo filters of the present invention are described in this paragraph, and some typical values and ranges of values for some time-frequency parameters are provided. This is demonstrated and compared by the model data: 1) a set of stereo filters to be matched, such as a typical stereo filter, and 2) a filter bank derived from the typical stereo filters by imposing simple compatibility. To obtain an oversimplified monophonic compatibility filter bank.

圖14A-14E顯示於該和及差濾波器響應中,在沿著該濾波器之長度的變化時間間隔處,當作頻率的一函數之能量的繪圖。雖然任意的,用於此敘述,本發明家選擇0-5毫秒、10-15毫秒、20-25毫秒、40-45毫秒、及80-85毫秒之時間片段。每一區段之5毫秒間隔係為比較功率位準維持一致之長度,且其係亦足夠捕捉該等濾波器中之可隨著時間的消逝而變稀疏之部份該回音及細節。圖14A-14E顯示根據本發明之一或更多態樣,在這些用於一典型對、用於過分簡化之單音相容性對、及用於新立體聲濾波器對的時間,用於5毫秒片段之頻譜。為決定這些繪圖,過分簡化之單音相容性對的脈衝響應係由該典型(待匹配對)所決定。再者,包括本發明之特色的濾波器之脈衝響應係根據上文所敘述之方法由該典型(待匹配對)所決定。。該頻 率能量響應係使用當作一短時窗口DFT之短時傅立葉轉換計算。沒有重疊被用於決定該五組之頻率響應。 Figures 14A-14E show plots of energy as a function of frequency at varying time intervals along the length of the filter in the sum and difference filter response. Although arbitrary, for this description, the inventors chose time segments of 0-5 milliseconds, 10-15 milliseconds, 20-25 milliseconds, 40-45 milliseconds, and 80-85 milliseconds. The 5 millisecond interval of each segment is a length that maintains a consistent power level, and is also sufficient to capture portions of the echo that are sparse as time elapses. 14A-14E show the time for a typical pair, for an oversimplified tone compatibility pair, and for a new stereo filter pair, in accordance with one or more aspects of the present invention, for 5 The spectrum of the millisecond segment. To determine these plots, the impulse response of an oversimplified pair of monophonic compatibility is determined by the typical (to be matched pair). Furthermore, the impulse response of the filter including the features of the present invention is determined by the typical (to be matched pair) according to the method described above. . The frequency The rate energy response is calculated using a short time Fourier transform as a short time window DFT. No overlap is used to determine the frequency response of the five groups.

注意所示濾波器可藉由一任意數量輕易地按比例變化,以致這些繪圖中所表達之值將以一相對及定量之意義解釋。所感興趣者不是該實際位準,而是當與該個別之和濾波器脈衝響應比較時,在該等個別差濾波器脈衝響應之頻譜的特別部份變得可忽略之時間。 Note that the filters shown can be easily scaled by any number such that the values expressed in these plots will be interpreted in a relative and quantitative sense. The person of interest is not the actual level, but the time at which a particular portion of the spectrum of the individual difference filter impulse responses becomes negligible when compared to the individual sum filter impulse response.

圖14A,用於在0毫秒時間開始之首先5毫秒,其能被看出該三個響應係幾乎完全相同的。這是該響應之很早期部份,其係基於來自一虛擬喇叭位置之HRTF,以賦予一方向感。於此時中,由於該遮蔽效應及支配的最初脈衝,該信號之任何散佈或該濾波器中之回音係大部份感覺被忽視的。 Figure 14A, for the first 5 milliseconds to start at 0 milliseconds, it can be seen that the three response systems are almost identical. This is a very early part of the response, based on the HRTF from a virtual horn location to give a sense of direction. At this point, due to the shadowing effect and the dominant initial pulse, any dispersion of the signal or the echo in the filter is mostly ignored.

於圖14B中,用於在10毫秒時間開始之5毫秒,用於該過分簡化方式之和信號係零。該和響應之稍後部份已被消去。相較之下,該新穎的濾波器對、例如上文所敘述中被決定者,仍然維持該和濾波器中之一些信號能量低於4kHz。所有三濾波器之差響應係類似的,使該新穎的濾波器對差脈衝響應在較高之頻率具有稍微更多的能量。 In Figure 14B, for 5 milliseconds starting at 10 milliseconds, the sum signal is zero for the oversimplified mode. The later part of the sum response has been eliminated. In contrast, the novel filter pair, such as those determined above, still maintains some of the signal energy in the sum filter below 4 kHz. The difference response of all three filters is similar, making the novel filter have slightly more energy for the differential impulse response at higher frequencies.

於圖14C中,用於在20毫秒時間開始之5毫秒,該新穎濾波器對之和濾波器係隨著該頻寬下降至約1kHz而進一步衰減。該新穎濾波器對之差濾波器被提升,以對於一典型或待匹配濾波器對整體維持一類似立體聲位準及頻率響應。 In Figure 14C, for 5 milliseconds starting at 20 milliseconds, the novel filter pair and filter system are further attenuated as the bandwidth drops to about 1 kHz. The novel filter pair difference filter is boosted to maintain a similar stereo level and frequency response for a typical or to-be-matched filter pair.

於圖14D中,用於在40毫秒時間開始之5毫秒,僅只保持該新穎濾波器對之和濾波器的最低分量。最後於圖14E中,用於在80毫秒開始之5毫秒,該過分簡化及新穎濾波器對中之和濾波器脈衝響應係可忽略的。 In Figure 14D, for 5 milliseconds starting at 40 milliseconds, only the lowest component of the novel filter pair and the filter is maintained. Finally, in Figure 14E, for 5 ms starting at 80 ms, the oversimplified and novel filter pair sum filter impulse response is negligible.

如此,一組立體聲濾波器被提出,具有該立體聲濾波器脈衝響應的一修飾,其被組構來達成非常好之單音播放相容性。於一些具體實施例中,該等濾波器被組構成使得該單音響應被限制於該首先40毫秒。 Thus, a set of stereo filters is proposed with a modification of the stereo filter impulse response that is organized to achieve very good monophonic playback compatibility. In some embodiments, the filters are grouped such that the tone response is limited to the first 40 milliseconds.

以下之性質有關用於達成良好立體聲響應及良好的單音播放相容性兩者之濾波器的有效性。於這些性質中,“濾波器範圍”及“濾波器長度”係該濾波器之脈衝響應掉落低於其最初值之-60分貝的地點。這在該技藝中亦已知為該“回響時間”。 The following properties relate to the effectiveness of filters for achieving good stereo response and good monophonic playback compatibility. Among these properties, "filter range" and "filter length" are the locations where the impulse response of the filter drops below -60 decibels of its original value. This is also known in the art as the "reverberation time".

以下之性質允許吾人區別在此中所敘述之本發明濾波器與其他立體聲濾波器及單音播放相容立體聲濾波器。 The following properties allow us to distinguish between the inventive filter described herein and other stereo filters and monophonic compatible stereo filters.

●該和及差濾波器實質上係不同的。用於一般之立體聲濾波器,該和及差濾波器橫越該時頻繪圖顯示類似之強度及衰減特徵。 • The sum and difference filters are essentially different. For general stereo filters, the sum and difference filters traverse the time-frequency plot to show similar intensity and attenuation characteristics.

●該和濾波器在所有頻率係比該差濾波器顯著地較短。雖然該和濾波器在用於典型收聽室期間中典型將稍微較短的,這並不是那麼顯著的。用於單音相容性,該和濾波器必需實質上較短的。 • The sum filter is significantly shorter than the difference filter at all frequencies. Although the sum filter will typically be slightly shorter during use in a typical listening room, this is not so significant. For tone compatibility, the sum filter must be substantially shorter.

●和濾波器在橫越不同頻率的長度中顯示一顯著之差異。這是與該過分簡化之方式相比較,在此該和濾波器在橫 越頻率的長度中係合理恆定的。 • The filter and the filter show a significant difference in the length across the different frequencies. This is compared to the way of oversimplification, where the sum filter is in the horizontal The length of the frequency is reasonably constant.

●該和濾波器在高頻係較短的及在低頻係較長的。 ● The sum filter is short in the high frequency system and long in the low frequency system.

注意一類似修飾可被達成,其中該總和頻道之抑制係更進取的(較佳之單音響應)、或更保守的(較佳之立體聲響應)。 Note that a similar modification can be achieved where the suppression of the sum channel is more aggressive (preferably monophonic response) or more conservative (better stereo response).

以更定量之術語,為達成立體聲響應及單音播放相容性的一良好結合,下文被發現為真實的: In a more quantitative term, the following is found to be true for a good combination of stereo response and monophonic playback compatibility:

差濾波器Difference filter

●例如在10kHz以上之差濾波器的高頻不會延伸超出大約10毫秒。於另一示範具體實施例中,大約20毫秒之差濾波器長度係仍然可接收的,而大約40毫秒之濾波器長度,一單音信號開始聽起來有回音的。 • For example, the high frequency of the difference filter above 10 kHz does not extend beyond about 10 milliseconds. In another exemplary embodiment, the difference in filter length of about 20 milliseconds is still acceptable, and for a filter length of about 40 milliseconds, a single tone signal begins to sound echo.

●例如在該差濾波器的3kHz及4kHz間之低頻係較長的,延伸出至大約40毫秒或約該差濾波器在該頻率之回響長度的1/8至1/4。 • For example, the low frequency between 3 kHz and 4 kHz of the difference filter is longer, extending to about 40 milliseconds or about 1/8 to 1/4 of the reverberation length of the difference filter at the frequency.

●甚至在較低頻率,比如說低於2kHz,該差濾波器在該最低頻率用於一非常好之響應應不比大約80毫秒較長。於一些具體實施例中,甚至120毫秒之長度聽起來可接收的,雖然用在少於2kHz具有大約160毫秒之濾波器長度,一單音信號開始聽起來有回音的。 • Even at lower frequencies, say below 2 kHz, the difference filter should not be used for a very good response at this lowest frequency for no more than about 80 milliseconds. In some embodiments, even a length of 120 milliseconds is audible, although with a filter length of less than 2 kHz having a waveform of about 160 milliseconds, a single tone signal begins to sound echoed.

再者,用於具有此強制性差濾波器之良好立體聲響應,該整個範圍、例如該差濾波器之回響應不會太長。本發明家已發現該200毫秒之回響時間產生優異之結果,400 毫秒產生可接收之結果,雖然該音頻具有800毫秒之濾波器長度時開始聽起來有問題。 Furthermore, for a good stereo response with this mandatory difference filter, the entire range, for example the back filter's back response, is not too long. The inventors have found that the reverberation time of 200 milliseconds produces excellent results, 400 The millisecond produces a receivable result, although the audio starts to sound problematic when it has a filter length of 800 milliseconds.

和濾波器And filter

表1提供一組用於不同頻帶的和濾波器脈衝響應長度之典型值、及亦提供用於該等頻帶之和濾波器脈衝響應長度的值之範圍,其仍然將提供單音播放相容性及收聽室空間化間之平衡。 Table 1 provides a set of values for the sum of the filter impulse response lengths for different frequency bands, and also provides a range of values for the sum of the filter impulse response lengths of the bands, which will still provide monophonic playback compatibility. And the balance between the room and the listening room.

選擇該時間相依頻率修飾視該想要之立體聲響應的自然及混響感而定,例如以一組如上文所敘述之待匹配立體聲濾波器hL0(t)及hR0(t)為其特徵,且亦視用於該單音混合頂抗該等立體聲濾波器中之近似值或限制中的透明度之優選而定。 Selecting the time-dependent frequency modifier depends on the natural and reverberant perception of the desired stereo response, for example, characterized by a set of stereo filters h L0 (t) and h R0 (t) to be matched as described above. And also depends on the preference for the transparency in the approximation or limit of the monophonic hybrid top filters.

為有利於藉由本發明所指示之和濾波器的修飾之敘述,該示範資料現在被呈現為該有關濾波器能量在時間及頻率之二維映射上方的繪圖。圖15A及15B在該時頻平面上顯示相等之衰減輪廓,分別用於一示範立體聲濾波器對具體實施例之和及頻率濾波器脈衝響應,而圖16A及16B 顯示該時頻繪圖、亦即頻譜圖之表面的等角視圖。該輪廓資料係藉由在5毫秒長區段上使用該窗口式短時傅立葉轉換所獲得,其分開地開始1.5毫秒,亦即其具有顯著之重疊。該等角視圖使用3毫秒之窗口長度,而沒有重疊,亦即資料每隔3毫秒開始。圖17A及17B顯示該時頻繪圖之表面的與圖16A及16B相同之等角視圖,但分別用於一典型立體聲濾波器對之和及頻率濾波器脈衝響應,特別地是,那些用於圖16A及16B之立體聲濾波器將匹配注意於一典型之立體聲濾波器對中,該和及差濾波器之個別脈衝響應的時頻繪圖之形狀係不是不同的。 To facilitate the description of the modification of the filter and the filter indicated by the present invention, the exemplary data is now presented as a plot of the filter energy over a two-dimensional map of time and frequency. 15A and 15B show equal attenuation profiles on the time-frequency plane, respectively, for an exemplary stereo filter versus the sum of the specific embodiment and the frequency filter impulse response, while FIGS. 16A and 16B An isometric view of the time-frequency plot, that is, the surface of the spectrogram is displayed. The profile data is obtained by using the windowed short-time Fourier transform over a 5 millisecond long segment, which starts 1.5 milliseconds separately, i.e., it has significant overlap. The isometric view uses a window length of 3 milliseconds without overlap, ie the data starts every 3 milliseconds. 17A and 17B show the same isosceles view of the surface of the time-frequency plot as in Figs. 16A and 16B, but for a typical stereo filter pair sum and frequency filter impulse response, respectively, in particular, those used for The stereo filters of 16A and 16B will match the attention of a typical stereo filter pair, and the shape of the time-frequency plot of the individual impulse responses of the sum and difference filters is not different.

注意該過分簡化之單音相容性濾波器對將顯示一和濾波器脈衝,對於所有頻率,其響應緊接及突然地下降至低於覺察得出之位準。 Note that this oversimplified monophonic compatibility filter pair will display a sum filter pulse, for all frequencies, the response is immediately followed and suddenly dropped below the perceived level.

注意該時頻資料的一些平滑化被進行,以產生圖15A、15B、16A、16B、17A、及17B,以便簡化該等圖面,以便不會使具有該等個別響應中之小細節變化的時頻特徵之特色變模糊。 Note that some smoothing of the time-frequency data is performed to produce Figures 15A, 15B, 16A, 16B, 17A, and 17B to simplify the drawings so as not to have small detail variations in the individual responses. The characteristics of the time-frequency feature are blurred.

應注意的是在此中所呈現的所有繪圖及曲線圖中所示之分貝位準係僅只在一相對比例,且如此不是該等濾波器之絕對特徵及所敘述之圖案。一熟諳此技藝者將能夠解釋這些圖面及它們所敘述之特徵,而不需保持正確之詳細位準、時間、及頻譜修飾。 It should be noted that the decibel levels shown in all of the plots and graphs presented herein are only in a relative scale, and thus are not the absolute features of the filters and the recited patterns. Those skilled in the art will be able to interpret these drawings and the features they describe without maintaining the correct level of detail, time, and spectral modification.

測試test

本發明家以數個型式之來源材料運行主觀之測試,使該修飾被界定於上面表1之“典型和濾波器長度”欄位中及待匹配之立體聲脈衝響應被給與為圖14A-14E之範例。該待匹配之脈衝響應具有200-300毫秒回響時間之立體聲響應,且對應於杜比耳機DH3立體聲濾波器。在此無統計上顯著之案例,其中於該測試中,該等主題更喜歡優於另一立體聲響應的一立體聲響應。然而,藉由用於所有被測試之來源材料的所有主題,該單音混合實質上被改善及一致較佳的。 The inventors performed subjective tests on several types of source materials such that the modifications were defined in the "Typical and Filter Length" fields of Table 1 above and the stereo impulse response to be matched was given as Figures 14A-14E. An example. The impulse response to be matched has a stereo response of 200-300 millisecond reverberation time and corresponds to a Dolby Headphone DH3 stereo filter. There are no statistically significant cases in which the subject prefers a stereo response that is superior to another stereo response. However, the mono mix is substantially improved and consistently better by all the themes for all of the source materials tested.

經過喇叭播放Played by the speaker

使用上述立體聲濾波器之方法及設備係不只可適用於立體聲耳機播放,但可應用至立體喇叭播放。當揚聲器係相鄰時,一收聽者的左及右耳之間於收聽時有串音、例如在一喇叭之輸出及最遠離該喇叭的耳朵間之串音。譬如,用於一對放置在收聽者前面之立體聲喇叭,串音意指該左耳由該右喇叭聽到聲音,且右耳亦由該左喇叭聽到聲音。與該等喇叭及該收聽者間之距離作比較,當該等喇叭係充分接近時,該串音本質上造成該收聽者聽到該二喇叭輸出之和。這本質上係與單音播放相同的。 The method and device using the above stereo filter are not only applicable to stereo earphone playback, but can be applied to stereo speaker playback. When the speakers are adjacent, a listener's left and right ears have crosstalk between them when listening, such as the output of a speaker and the crosstalk between the ear farthest from the speaker. For example, for a pair of stereo speakers placed in front of the listener, crosstalk means that the left ear hears the sound from the right speaker, and the right ear also hears the sound from the left speaker. Comparing with the distance between the speakers and the listener, when the speakers are sufficiently close, the crosstalk essentially causes the listener to hear the sum of the outputs of the two speakers. This is essentially the same as mono playback.

實施該等濾波器Implementing these filters

再者,那些熟諳該技藝者將了解該等數位濾波器可被很多方法所實施。譬如,該等數位濾波器可被有限脈衝響 應(FIR)實施、該頻域中之實施、重疊轉換方法等所進行。很多此等方法係習知的,且如何應用它們至在此中所敘述之實施對於那些熟諳該技藝者將是易懂的。 Moreover, those skilled in the art will appreciate that such digital filters can be implemented in a number of ways. For example, these digital filters can be strobed with limited pulses. It is performed by (FIR) implementation, implementation in the frequency domain, overlap conversion method, and the like. Many of these methods are well known and how to apply them to the implementations described herein will be readily apparent to those skilled in the art.

注意其將被那些熟諳此技藝者所了解,即該等上面濾波器敘述未說明所有必需之零組件,諸如音頻放大器、及其他類似元件,且一熟諳此技藝者將已知加入此等元件,而不需進一步教導。再者,該等上面之實施係用於數位濾波。因此,用於類比輸入,那些熟諳該技藝者將了解類比至數位轉換器被包括在內。再者,數位至類比(D/A)轉換器將被了解為使用於將該數位信號輸出轉換成類比輸出,用於經過耳機播放,或於該音頻濾波案例中經過揚聲器。 It will be appreciated that those skilled in the art will appreciate that the above filter descriptions do not describe all of the necessary components, such as audio amplifiers, and the like, and those skilled in the art will be known to incorporate such components. Without further instruction. Again, these above implementations are for digital filtering. Therefore, for analog input, those skilled in the art will appreciate that analog to digital converters are included. Furthermore, a digital to analog (D/A) converter will be understood to be used to convert the digital signal output to an analog output for playback via headphones or through a speaker in the audio filtering case.

圖18顯示一音頻處理設備之實施形式,用於根據本發明之態樣處理一組音頻輸入信號。該音頻處理系統包括:一輸入介面方塊1821,其包括一被組構成將類比輸入信號轉換成對應之數位信號的類比至數位(A/D)轉換器;及一輸出方塊1823,其具有一數位至類比(D/A)轉換器,以將該被處理之信號轉換成類比輸出信號。於另一具體實施例中,該輸入方塊1821亦或取代該A/D轉換器包括一SPDIF(索尼/飛立普數位互連格式)介面,其被組構成除了類比輸入信號以外或非類比輸入信號而接收數位輸入信號。該設備包括一數位信號處理器(DSP)裝置1800,其能夠處理該輸入,以充分快速地產生該輸出。於一具體實施例中,該DSP裝置包括呈串列埠1817之形式的介面電路系統,該串列埠被組構成與該A/D及D/A轉換器資訊通訊 ,而沒有處理器負擔,且於一具體實施例中,一正常關閉元件記憶體1803及DMA(直接記憶體存取)引擎1813可由該晶片外記憶體1803拷貝資料至一晶片上記憶體1811,而不會干擾該輸入/輸出處理之操作。於一些具體實施例中,用於實施本發明在此中所敘述之態樣的程式碼可為在該晶片外記憶體1803中,且如所需地被載入至該晶片上記憶體1811。所示該DSP設備包括一程式記憶體1807,其包括造成該DSP設備之處理器部份1805實施在此中所敘述之濾波的程式碼1809。一外部匯流排多工器1815被包括用於需要該外部記憶體1803之案例。 Figure 18 shows an embodiment of an audio processing device for processing a set of audio input signals in accordance with aspects of the present invention. The audio processing system includes an input interface block 1821 including an analog to digital (A/D) converter configured to convert an analog input signal into a corresponding digital signal; and an output block 1823 having a digit An analog to analog (D/A) converter to convert the processed signal into an analog output signal. In another embodiment, the input block 1821 also includes or replaces the A/D converter with an SPDIF (Sony/Feilip Digital Interconnect Format) interface, which is grouped to form an analog input signal or a non-analog input. The signal receives the digital input signal. The device includes a digital signal processor (DSP) device 1800 that is capable of processing the input to produce the output sufficiently quickly. In a specific embodiment, the DSP device includes an interface circuit system in the form of a serial port 1817, the serial port is configured to communicate with the A/D and D/A converters. There is no processor burden, and in a specific embodiment, a normal shutdown component memory 1803 and a DMA (direct memory access) engine 1813 can copy data from the off-chip memory 1803 to a memory on the chip 1811. It does not interfere with the operation of this input/output processing. In some embodiments, the code for implementing the aspects of the invention described herein can be in the off-chip memory 1803 and loaded onto the on-wafer memory 1811 as desired. The illustrated DSP device includes a program memory 1807 that includes a code 1809 that causes the processor portion 1805 of the DSP device to perform the filtering described herein. An external bus multiplexer 1815 is included for the case where the external memory 1803 is required.

注意該晶片外及晶片上一詞不應被解釋為暗指有超過一個所顯示之晶片。於現代之應用中,所示DSP裝置1800方塊可隨同其他電路系統被提供當作待包括於一晶片中之“核心”。再者,那些熟諳該技藝者將了解圖18所示設備純粹係一範例。 Note that the words outside the wafer and on the wafer should not be construed as implying that there is more than one wafer shown. In modern applications, the illustrated DSP device 1800 blocks can be provided along with other circuitry as a "core" to be included in a wafer. Moreover, those skilled in the art will appreciate that the device shown in Figure 18 is purely an example.

同樣地,圖19A顯示一立體聲化設備之具體實施例的簡化方塊圖,該設備被組構成以針對經過前面喇叭播放的左、中心及右信號、及針對經由後方喇叭播放之左側環繞與右側環繞信號之形式接收五頻道之音頻資訊。該立體聲化器實施用於每一輸入之立體聲濾波器對,包括用於本發明之態樣的左側環繞及右側環繞信號,以致一經過耳機收聽之收聽者體驗空間內容,同時一收聽單音混合之收聽者以一愜意方式體驗該等信號,好像來自一單音來源。該立體聲化器係使用一處理系統1903、例如一包括DSP裝 置之系統實施,該DSP裝置包括至少一處理器1905。一記憶體1907被包括用於以指令之形式保持程式碼,且進一步能保持任何需要之參數。當執行時,該程式碼造成該處理系統1903執行濾波,如上文所敘述者。 Similarly, Figure 19A shows a simplified block diagram of a particular embodiment of a stereo device that is configured to target left, center, and right signals that are played through the front speakers, and to the left and right surrounds that are played through the rear speakers. The form of the signal receives the audio information of the five channels. The stereoizer implements a pair of stereo filters for each input, including left side surround and right surround signals for aspects of the present invention such that a listener listening through the headphones experiences spatial content while listening to a single tone mix The listener experiences the signals in a pleasing manner, as if from a single source. The stereoizer uses a processing system 1903, such as a DSP package In a system implementation, the DSP device includes at least one processor 1905. A memory 1907 is included for maintaining the code in the form of instructions and further capable of maintaining any desired parameters. When executed, the code causes the processing system 1903 to perform filtering, as described above.

同樣地,圖19B顯示一立體聲化設備之具體實施例的簡化方塊圖,該設備以針對經過前面喇叭播放的左及右前側信號、及針對經由後方喇叭播放之左後方與右後方信號之形式接收四頻道之音頻資訊。該立體聲化器實施用於每一輸入之立體聲濾波器對,包括用於本發明之態樣的左及右信號、與用於該左後方與右後方信號,以致一經過耳機收聽之收聽者體驗空間內容,同時一收聽單音混合之收聽者以一愜意方式體驗該等信號,好像來自一單音來源。該立體聲化器係使用一例如包括DSP裝置之處理系統1903實施,該DSP裝置具有一處理器1905。一記憶體1907被包括用於以指令之形式保持程式碼1909,且進一步能保持任何需要之參數。當執行時,該程式碼造成該處理系統1903執行濾波,如上文所敘述者。 Similarly, Figure 19B shows a simplified block diagram of a particular embodiment of a stereoizing device that receives signals for left and right front side signals that are played through the front speakers and for left and right rear signals that are played through the rear speakers. Audio information of four channels. The stereoizer implements a pair of stereo filters for each input, including left and right signals for the aspects of the present invention, and for the left rear and right rear signals such that a listener experience is heard through the headphones The spatial content, while listening to the monophonic listener, experiences the signals in a pleasant way, as if from a single source. The stereoizer is implemented using a processing system 1903, for example, including a DSP device having a processor 1905. A memory 1907 is included for holding the code 1909 in the form of instructions and further capable of maintaining any desired parameters. When executed, the code causes the processing system 1903 to perform filtering, as described above.

於一具體實施例中,以例如一組指令之程式邏輯組構電腦可讀取媒體,當藉由至少一處理器所執行時,該組指令造成實行在此中所敘述之方法的一組方法步驟。 In one embodiment, the computer readable medium is organized by, for example, a set of instructions, and when executed by at least one processor, the set of instructions causes a set of methods to perform the methods described herein. step.

除非另外特別地陳述,如由以下之討論變得明顯,應了解遍及利用諸如“處理”、“估算”、“計算”、“決定”等術語之說明書討論,意指一電腦或計算系統、或類似電子計算裝置之作用及/或處理,其將代表為物理、諸 如電子、參量之資料處理及/或轉換成同樣地代表為物理參量之另一資料。 Unless otherwise specifically stated, as will become apparent from the following discussion, it should be understood that a discussion of the terms, such as "processing," "estimating," "calculating," "decision," and the like, means a computer or computing system, or Similar to the role and/or processing of electronic computing devices, which will be represented as physics, Data such as electronics and parameters are processed and/or converted into another material that is equally represented as a physical parameter.

以一類似方式,該“處理器”一詞可意指任何裝置或一裝置的一部份,其處理例如來自暫存器及/或記憶體之電子資料,以將該電子資料轉換成另一例如可被儲存於暫存器及/或記憶體中之電子資料。一“電腦”或一“計算機”或一“計算平臺”可包括至少一處理器。 In a similar manner, the term "processor" may mean any device or portion of a device that processes, for example, electronic data from a register and/or memory to convert the electronic data into another For example, electronic data that can be stored in a scratchpad and/or memory. A "computer" or a "computer" or a "computing platform" can include at least one processor.

注意當一方法被敘述時,該方法包括數個元件,例如數個步驟,除非特別陳述,不隱含此等元件之任何排序、例如步驟之排序。 Note that when a method is recited, the method includes several elements, such as a number of steps, and unless otherwise stated, any ordering of such elements, such as the ordering of the steps, is not implied.

於一具體實施例中,在此中所敘述之方法論係可藉由一或更多處理器施行的,該等處理器接收具體化在一或更多電腦可讀取媒體上之電腦可執行(亦稱為機械可執行)的程式邏輯。該程式邏輯包括一組指令,其當藉由一或更多該等處理器所執行時,實行在此中所敘述之方法的至少一方法。任何能夠執行一組指定將採取之作用的指令(連續或別樣的)之處理器被包括在內。如此,一範例係一典型之處理系統,其包括一處理器或超過之處理器。每一處理器可包括中央處理系統、圖形處理單元、及可程式化DSP單元的一或多個。該處理系統另可包括一儲存子系統,其包括一記憶體子系統,該記憶體子系統包括主要RAM及/或靜態RAM、及/或ROM。該儲存子系統可另包括一或多個其他儲存裝置。一匯流排子系統可被包括,用於與於該等零組件之間通訊。該處理系統另可為一分散式處理 系統,具有藉由一網路所耦接之處理器。如果該處理系統需要一顯示器,此一顯示器可被包括,例如一液晶顯示器(LCD)、有機發光顯示器、電漿顯示器、陰極射線管(CRT)顯示器等。如果手動資料輸入係需要的,該處理系統亦諸如包括一輸入設備,諸如鍵盤之文數字輸入單元、諸如滑鼠之指向控制裝置等的一或多個。如在此中所使用之單元,如果由該上下文清楚及除非別樣明確地陳述,儲存裝置、儲存子系統等術語亦涵蓋諸如碟片驅動器單元之儲存裝置。於一些架構中,該處理系統可包括一聲音輸入裝置、及一網路介面裝置。該儲存子系統如此包括一承載程式邏輯(例如軟體)之電腦可讀取媒體,該程式邏輯包括一組指令,以當藉由一或多個處理器所執行時,造成實行在此中所敘述之方法的一或多個方法。該程式邏輯可於其藉由該處理系統執行期間常駐在一硬碟機中,或亦可完全地或至少局部地常駐在該RAM內及/或該處理器內。如此,該記憶體及該處理器亦構成電腦可讀取媒體,在其上者係被編碼之程式邏輯,例如呈指令之形式。 In one embodiment, the methodology described herein can be performed by one or more processors that receive computer executables embodied on one or more computer readable media ( Also known as mechanical executable) program logic. The program logic includes a set of instructions that, when executed by one or more of the processors, perform at least one of the methods described herein. Any processor capable of executing a set of instructions (continuous or otherwise) that specify the role to be taken is included. Thus, an example is a typical processing system that includes a processor or a processor. Each processor can include one or more of a central processing system, a graphics processing unit, and a programmable DSP unit. The processing system can further include a storage subsystem including a memory subsystem including a primary RAM and/or static RAM, and/or a ROM. The storage subsystem may additionally include one or more other storage devices. A busbar subsystem can be included for communication with the components. The processing system can also be a decentralized process The system has a processor coupled by a network. If the processing system requires a display, the display can include, for example, a liquid crystal display (LCD), an organic light emitting display, a plasma display, a cathode ray tube (CRT) display, and the like. If manual data entry is required, the processing system also includes, for example, one or more input devices, such as an alphanumeric input unit for a keyboard, a pointing control device such as a mouse. As used herein, the terms "storage device, storage subsystem" and the like also encompass a storage device such as a disc drive unit, if it is clear from the context and unless otherwise explicitly stated. In some architectures, the processing system can include a voice input device and a network interface device. The storage subsystem thus includes a computer readable medium carrying program logic (e.g., software), the program logic including a set of instructions to cause execution as described herein when executed by one or more processors One or more methods of the method. The program logic may reside in a hard disk drive during execution by the processing system, or may reside entirely or at least partially within the RAM and/or within the processor. Thus, the memory and the processor also constitute a computer readable medium on which the programmed logic is encoded, for example in the form of instructions.

再者,一電腦可讀取媒體可形成、或被包括在一電腦程式產品中。 Furthermore, a computer readable medium can be formed or included in a computer program product.

於另一選擇具體實施例中,該一或多個處理器操作為一獨立的裝置,或於一網路式部署中,可被例如網路連接至其他處理器,該一或多個處理器可在伺服者-客戶端網路環境中之伺服器或客戶端機器的能力中操作,或操作為一點對點或分散式網路環境中之個別系統。該一或多個處 理器可形成一個人電腦(PC)、平板PC、機上盒(STB)、個人數位助理(PDA)、行動電話、網絡計算機、網路路由器、開關或橋接器、或任何能夠執行一組指令(連續或別樣的)之機器,該組指令指定將藉由那機器所採取之作用。 In another optional embodiment, the one or more processors operate as a separate device, or in a networked deployment, can be connected to other processors, such as a network, the one or more processors It can operate in the capabilities of a server or client machine in a server-client network environment, or as an individual system in a peer-to-peer or decentralized network environment. One or more places The processor can form a personal computer (PC), tablet PC, set-top box (STB), personal digital assistant (PDA), mobile phone, network computer, network router, switch or bridge, or any device capable of executing a set of instructions ( For continuous or different machines, the set of instructions specifies the role to be taken by that machine.

注意雖然一些圖解僅只顯示承載包括指令之邏輯的單一處理器及單一記憶體,那些熟諳該技藝者將了解上述許多零組件被包括,但不明確地顯示或敘述,以便不會使本發明之態樣難理解。譬如,雖然僅只單一機器被說明,該“機器”一詞亦將被視為包括機器之任何集合,該等機器個別地或共同地執行一組(或多組)指令,以實行在此中所討論之方法論的任何一個或多個。 Note that while some of the diagrams only show a single processor and a single memory that carries the logic including instructions, those skilled in the art will appreciate that many of the above-described components are included, but are not explicitly shown or described so as not to obviate the present invention. It's hard to understand. For example, although only a single machine is illustrated, the term "machine" will also be taken to include any collection of machines that individually or collectively execute a set (or sets) of instructions for execution therein. Discuss any one or more of the methodology.

如此,在此中所敘述之每一方法的一具體實施例係呈一電腦可讀取媒體之形式,並以一組指令組構,例如被用於在一或多個處理器上執行之電腦程式,例如為信號處理設備的部件之一或多個處理器。如此,如將被那些熟諳此技藝者所了解,本發明之具體實施例可被具體化為一方法、一諸如特別用途設備之設備、一諸如資料處理系統之設備、或一例如電腦程式產品之電腦可讀取媒體。該電腦可讀取媒體承載包括一組指令之邏輯,當在一或多個處理器上執行時,該組指令造成實行方法步驟。據此,本發明之態樣可採取一方法、一完全硬體具體實施例、一完全軟體具體實施例、或一結合軟體及硬體態樣的具體實施例之形式。再者,本發明可採取程式邏輯之形式,例如於一電腦可讀取媒體中,例如一電腦可讀取儲存媒體上之電腦程式 、或以電腦可讀取程式碼組構之電腦可讀取媒體,例如一電腦程式產品。 Thus, a specific embodiment of each of the methods described herein is in the form of a computer readable medium and is organized by a set of instructions, such as a computer for execution on one or more processors. The program is, for example, one of a component of a signal processing device or a plurality of processors. Thus, as will be appreciated by those skilled in the art, the embodiments of the present invention can be embodied in a method, a device such as a special purpose device, a device such as a data processing system, or a computer program product. The computer can read the media. The computer readable media bearer includes logic for a set of instructions that, when executed on one or more processors, cause the method steps to be performed. Accordingly, aspects of the invention may be in the form of a method, a complete hardware embodiment, a complete software embodiment, or a combination of software and hardware aspects. Furthermore, the present invention can take the form of program logic, such as in a computer readable medium, such as a computer readable storage computer program Or a computer readable medium that can be read by a computer readable code, such as a computer program product.

雖然在一示範具體實施例中,該電腦可讀取媒體被顯示為單一媒體,該“媒體”一詞應被視為包括單一媒體或多數媒體(例如一集中或分散式資料庫、及/或相關快取記憶體與伺服器),其儲存該一或多組指令。該“電腦可讀取媒體”一詞亦將被視為包括任何電腦可讀取媒體,其係能夠以藉由一或多個處理器所執行之一組指令儲存、編碼、或以別的方式組構,且造成本發明之方法論的任何一個或多個之實行。一電腦可讀取媒體可採取很多形式,包括、但不限於不變性媒體及易變性媒體。不變性媒體包括譬如光碟、磁碟、及磁光碟。易變性媒體包括動態記憶體、諸如主記憶體。 Although in a particular embodiment the computer readable medium is displayed as a single medium, the term "media" should be taken to include a single medium or a plurality of media (eg, a centralized or decentralized database, and/or A related cache memory and server) that stores the one or more sets of instructions. The term "computer readable medium" will also be taken to include any computer readable medium that can be stored, encoded, or otherwise stored in a set of instructions executed by one or more processors. It is organized and results in the implementation of any one or more of the methodology of the present invention. A computer readable medium can take many forms, including, but not limited to, immutable media and volatile media. Immutable media include, for example, compact discs, magnetic disks, and magneto-optical disks. Volatile media includes dynamic memory, such as main memory.

在一具體實施例中,將了解所討論的方法之步驟係藉由執行儲存器中所儲存之指令的處理系統(例如電腦系統)之適當處理器(或各處理器)所施行。亦將了解本發明之具體實施例係不限於任何特別之工具或程式規劃技術,且本發明可使用任何用於實施在此中所敘述之功能性的適當技術被實施。再者,具體實施例係不限於任何特別之程式規劃語言或作業系統。 In one embodiment, it will be appreciated that the steps of the method discussed are performed by a suitable processor (or processor) of a processing system (e.g., a computer system) that executes instructions stored in a memory. It will also be appreciated that the specific embodiments of the present invention are not limited to any particular tool or programming technique, and that the invention can be implemented using any suitable technique for implementing the functionality described herein. Moreover, the specific embodiments are not limited to any particular programming language or operating system.

遍及此說明書,參考“一具體實施例”或“具體實施例”意指關於該具體實施例所敘述之特別的特色、結構或特徵被包括於本發明之至少一具體實施例中。如此,在遍及此說明書之各種位置中,“一具體實施例”或“具體實 施例”詞組之狀態係不須全部參考相同之具體實施例,但可能全部參考相同之具體實施例。再者,該特別之特色、結構或特徵可被以任何合適之方式組合,如對於普通熟諳該技藝者將由此揭示內容於一或多個具體實施例中變得明顯。 Throughout the specification, reference to "a particular embodiment" or "an embodiment" means that a particular feature, structure, or feature described in connection with the particular embodiment is included in at least one embodiment of the invention. Thus, in various locations throughout the specification, "a particular embodiment" or "specifically The singularity of the phrase "a" or "an" It will be apparent to those skilled in the art that this disclosure will become apparent in one or more embodiments.

同樣地應了解於本發明之範例具體實施例的上面敘述中,本發明之各種特色有時候被一起組織在單一具體實施例、圖面、或其敘述中,用於簡化該揭示內容及輔助各種發明態樣之一或多個的理解之目的。然而,此揭示內容之方法不被解釋為反映一用意,即所申請之發明比在每一申請專利範圍中所明確引述者需要更多特色。反之,如以下之申請專利範圍所反映,本發明之態樣在於少於單一先前揭示具體實施例之所有特色。如此,在示範具體實施例的敘述之後,該等申請專利範圍據此明確地併入此示範具體實施例之敘述,使每一申請專利範圍獨自當作本發明的一分開之具體實施例。 It is to be understood that in the foregoing description of the exemplary embodiments of the invention, the various features of the invention are The purpose of understanding one or more of the inventive aspects. However, the method of this disclosure is not to be interpreted as reflecting the intention that the claimed invention requires more features than those specifically recited in the scope of each application. On the contrary, the invention is characterized by less than all features of a single previously disclosed embodiment. Thus, the scope of the present invention is to be construed as being limited to the specific embodiments of the invention.

再者,雖然在此中所敘述之一些具體實施例包括一些特色,但無其他特色被包括於其他具體實施例中,不同具體實施例之特色的結合係意指在本發明之範圍內,且形成不同具體實施例,如將被那些熟諳該技藝者所了解。譬如,於以下之申請專利範圍中,所申請之具體實施例的任一個能夠被以任何組合使用。 Furthermore, although some specific embodiments described herein include some features, nothing else is included in the other specific embodiments, and combinations of features of different specific embodiments are intended to be within the scope of the present invention. Different specific embodiments are formed as will be appreciated by those skilled in the art. For example, in the following patent claims, any of the specific embodiments of the application can be used in any combination.

再者,部份該等具體實施例在此中被敘述為一方法或一方法之各元件的組合,其能被一電腦系統之處理器或藉 由實行該功能之其他機構所實行。如此,一具有用於實行此一方法或一方法之要素的所需指令之處理器形成一用於實行該方法或一方法之要素的機構。再者,一設備具體實施例的在此中所敘述之元件係一機構之範例,用於實行藉由該元件所施行之功能,該元件用於實行本發明之目的。 Furthermore, some of the specific embodiments are described herein as a combination of components of a method or a method that can be Implemented by other agencies that perform this function. Thus, a processor having the required instructions for implementing the elements of the method or method forms a mechanism for implementing the elements of the method or method. Furthermore, the elements of a device embodiment described herein are examples of a mechanism for performing the functions performed by the element for performing the purposes of the present invention.

於在此中所提供之敘述中,極多特定之細節被提出。然而,應了解本發明之具體實施例可沒有這些特定之細節地被實踐。於其他情況中,熟知之方法、結構及技術未被詳細地顯示,以便不會使此敘述之理解模糊。 In the narratives provided herein, a number of specific details are presented. However, it is understood that the specific embodiments of the invention may be practiced without these specific details. In other instances, well-known methods, structures, and techniques have not been shown in detail so as not to obscure the understanding of the description.

如在此中所使用,除非以別的方式指定該序數詞“第一”、“第二”、“第三”等之使用敘述一共同物件,僅只指示所參考之相像物件的不同情況,且不被意欲暗指如此敘述之物件必需為暫時地、空間地、排列地、或以任何其他方式在一給定順序中。 As used herein, unless the use of the ordinal numerals "first", "second", "third", etc., is used to describe a common item, only the different instances of the referenced object are referred to, and It is not intended to imply that the items so recited must be temporarily, spatially, arranged, or in any other manner in a given order.

此說明書中之先前技藝的任何討論將絕不被考慮為一項表達,即此先前技藝係普遍認知、公開已知、或形成該領域中之常識的一部份。 Any discussion of prior art in this specification will in no way be considered as an expression that the prior art is generally recognized, publicly known, or forms part of the general knowledge in the field.

於下面之申請專利範圍及在此中之敘述中,包括、由...所組成、或其包括等詞的任一個係一開放術語,其意指包括隨後之至少該等元件/特色,但不排除其他者。如此,當使用於該等申請專利範圍時,包括一詞將不被解釋為限定於此後列出之機構或元件或步驟。譬如,包括A及B之裝置的表達之範圍將不被限制於僅只由元件A及B所組成之裝置。如在此中所使用,包括或其包括或該包括等 詞的任一個係亦一開放術語,其亦意指包括該術語之後的至少該等元件/特色,但不排除其他者。如此,包括係與涵括同義及意指涵括。 In the following claims, and in the context of the following claims, any of the terms including, consisting of, or the like, is an open term, which is meant to include at least such elements/features, but Others are not excluded. As such, the use of the term "comprising" is not to be construed as a limitation For example, the scope of expression of devices including A and B will not be limited to devices consisting only of components A and B. As used herein, including or including or including Any of the words are also open terms, which are also intended to include at least such elements/features after the term, but do not exclude others. Thus, the inclusion and the meaning are included in the meaning and meaning.

同樣地,當使用於該等申請專利範圍中時,其將被察見耦接一詞不應被解釋為僅只受限於直接之連接。該“耦接”及“連接”等詞隨著其衍生詞可被使用。應了解這些術語係不意欲彼此為同義詞。如此,一裝置A耦接至一裝置B的表達之範圍不應被限制於裝置或系統,其中裝置A之輸出係直接連接至一裝置B之輸入。其意指於A之輸出及B的輸入之間存在有一路徑,該路徑可為一包括其他裝置或機構之路徑。“耦接”可意指該二或更多元件係呈直接物理或電接觸,或該二或更多元件未彼此直接接觸,但又仍然彼此合作或相互作用。 Likewise, when used in the scope of such claims, it will be understood that the term "coupled" is not to be construed as limited only to the direct connection. The terms "coupled" and "connected" can be used with their derivatives. It should be understood that these terms are not intended to be synonymous with each other. Thus, the scope of expression of a device A coupled to a device B should not be limited to a device or system, where the output of device A is directly connected to the input of a device B. It means that there is a path between the output of A and the input of B, which may be a path including other devices or mechanisms. "Coupled" may mean that the two or more elements are in direct physical or electrical contact, or that the two or more elements are not in direct contact with each other, but still cooperate or interact with each other.

如此,雖然已在此敘述吾人相信為本發明之較佳具體實施例者,那些熟諳此技藝者將認知可對其作成其他及進一步之修改,而未由本發明之精神脫離,且其係意欲主張所有此等變化及修改如落在本發明之範圍內。譬如,在上面所給與之任何公式係僅只可被使用之程序的代表性者。功能性可被加入該等方塊圖或由該等方塊圖刪除,且操作可在功能方塊之中被交換。步驟可被加入在本發明的範圍內所敘述之方法或由其刪除。 Having thus described the preferred embodiments of the present invention, those skilled in the art will recognize that other and further modifications can be made thereto without departing from the spirit of the invention, and All such changes and modifications are within the scope of the invention. For example, any of the formulas given above are representative of only the programs that can be used. Functionality may be added to or deleted from the block diagrams and operations may be exchanged among the functional blocks. The steps can be added to or deleted from the methods described within the scope of the invention.

101‧‧‧立體聲化器 101‧‧‧Stereostat

103‧‧‧立體聲濾波器 103‧‧‧Stereo Filter

104‧‧‧立體聲濾波器 104‧‧‧Stereo Filter

105‧‧‧耳機 105‧‧‧ headphone

107‧‧‧收聽者 107‧‧‧ Listeners

109‧‧‧虛擬喇叭 109‧‧‧Virtual Horn

203-1‧‧‧立體聲化器 203-1‧‧‧Stereophone

203-2‧‧‧立體聲化器 203-2‧‧‧Stereophone

203-M‧‧‧立體聲化器 203-M‧‧‧Stereophone

204-1‧‧‧立體聲化器 204-1‧‧‧Stereophone

204-2‧‧‧立體聲化器 204-2‧‧‧Stereostat

204-M‧‧‧立體聲化器 204-M‧‧‧Stereophone

205‧‧‧加法器 205‧‧‧Adder

206‧‧‧加法器 206‧‧‧Adder

209-1‧‧‧喇叭 209-1‧‧‧ Speaker

209-2‧‧‧喇叭 209-2‧‧‧ Speaker

209-Mν‧‧‧喇叭 209-M ν ‧‧‧ Speaker

303‧‧‧立體聲化器 303‧‧‧Stereostat

305‧‧‧下混頻器 305‧‧‧Down Mixer

307‧‧‧濾波器 307‧‧‧ filter

308‧‧‧濾波器 308‧‧‧ filter

401‧‧‧混洗器 401‧‧‧Breaker

403‧‧‧和濾波器 403‧‧‧ and filter

404‧‧‧差濾波器 404‧‧‧Differential filter

405‧‧‧解混洗器 405‧‧‧Unmixer

603‧‧‧混洗器 603‧‧‧Breaker

605‧‧‧和濾波器 605‧‧‧ and filter

607‧‧‧差時變濾波器 607‧‧‧Differential time-varying filter

609‧‧‧解混洗器 609‧‧‧Unmixer

1800‧‧‧數位信號處理器裝置 1800‧‧‧Digital Signal Processor Unit

1803‧‧‧正常關閉元件記憶體 1803‧‧‧Normal shutdown of component memory

1805‧‧‧處理器部份 1805‧‧‧ Processor section

1807‧‧‧程式記憶體 1807‧‧‧Program memory

1809‧‧‧程式碼 1809‧‧‧ Code

1811‧‧‧晶片上記憶體 1811‧‧‧ Memory on the wafer

1813‧‧‧直接記憶體存取引擎 1813‧‧‧Direct Memory Access Engine

1815‧‧‧外部匯流排多工器 1815‧‧‧External bus multiplexer

1817‧‧‧串列埠 1817‧‧‧Chain

1821‧‧‧輸入介面方塊 1821‧‧‧Input interface box

1823‧‧‧輸出方塊 1823‧‧‧ Output Blocks

1903‧‧‧處理系統 1903‧‧‧Processing system

1905‧‧‧處理器 1905‧‧‧ Processor

1907‧‧‧記憶體 1907‧‧‧ memory

1909‧‧‧程式碼 1909‧‧‧Code

圖1顯示包括一對立體聲濾波器之立體聲化器的簡化 方塊圖,該對立體聲濾波器用於處理單一輸入信號與包括本發明的一具體實施例。 Figure 1 shows a simplified version of a stereoizer that includes a pair of stereo filters. A block diagram of a pair of stereo filters for processing a single input signal and including a particular embodiment of the present invention.

圖2顯示包括一或多對立體聲濾波器的立體聲化器之簡化方塊圖,該等立體聲濾波器用於對應於一或多個輸入信號作處理及包括本發明的一具體實施例。 2 shows a simplified block diagram of a stereoizer including one or more pairs of stereo filters for processing corresponding to one or more input signals and including a particular embodiment of the present invention.

圖3顯示一立體聲化器之簡化方塊圖,該立體聲化器具有一或多個音頻輸入信號及產生左耳與右耳輸出信號及可包括本發明的一具體實施例,該等輸出信號被混合成一單音混合。 3 shows a simplified block diagram of a stereoizer having one or more audio input signals and generating left and right ear output signals and may include a specific embodiment of the present invention, the output signals being mixed into one Monophonic mixing.

圖4A顯示藉由根據一對立體聲濾波器的和及差之後的混洗操作,隨後有一解混洗(de-shuffling)操作,其可包括本發明的一具體實施例。 Figure 4A shows a de-shuffling operation followed by a shuffling operation following the sum and difference of a pair of stereo filters, which may include a particular embodiment of the present invention.

圖4B顯示一在左及右輸入信號上之混洗操作,隨後有一解混洗操作,該等輸入信號代表可包括本發明的一具體實施例之立體聲濾波器的脈衝響應。 Figure 4B shows a shuffling operation on the left and right input signals followed by a deshuffling operation representative of the impulse response of a stereo filter that may include an embodiment of the present invention.

圖5顯示一示範之立體聲濾波器脈衝響應。 Figure 5 shows an exemplary stereo filter impulse response.

圖6顯示信號處理設備具體實施例之簡化方塊圖,其在一對代表其立體聲化性質將被匹配的立體聲濾波器脈衝響應之輸入信號上操作。該處理設備被組構成輸出代表立體聲濾波器脈衝響應之信號,該等脈衝響應能夠根據本發明之一或更多態樣立體聲化及產生一自然聲之單音混合。 Figure 6 shows a simplified block diagram of a particular embodiment of a signal processing device operating on a pair of input signals representative of the stereo filter impulse response whose stereogenic properties are to be matched. The processing device is configured to output a signal representative of a stereo filter impulse response that is capable of stereophoning and producing a natural sound monophonic mixture in accordance with one or more aspects of the present invention.

圖7顯示操作諸如圖6之信號處理設備以產生立體聲脈衝響應的方法之具體實施例的簡化流程圖。 7 shows a simplified flow diagram of a particular embodiment of a method of operating a signal processing device such as that of FIG. 6 to produce a stereo impulse response.

圖8顯示MATLAB(麻薩諸塞州內迪克市之 Mathworks公司)語法中之編碼的一部份,其實行將一對代表立體聲濾波器脈衝響應之信號轉換成代表被修改之立體聲濾波器的脈衝響應之信號的方法具體實施例。 Figure 8 shows MATLAB (Neidick, MA) A portion of the encoding in the grammar of Mathworks, which implements a method of converting a pair of signals representative of the stereo filter impulse response into a signal representative of the impulse response of the modified stereo filter.

圖9顯示一使用於圖6之設備具體實施例及圖7的方法具體實施例之時變濾波器的脈衝響應、對在一組不同時刻之每一時刻的脈衝之繪圖。 Figure 9 shows an impulse response of a time varying filter for use with the apparatus embodiment of Figure 6 and the method embodiment of Figure 7, for plotting pulses at each of a set of different times.

圖10顯示一使用於圖6之設備具體實施例及圖7的方法具體實施例之時變濾波器的頻率響應振幅在一組不同時刻之每一時刻的繪圖。 Figure 10 shows a plot of the frequency response amplitude of a time varying filter used in a particular embodiment of the apparatus of Figure 6 and the method embodiment of Figure 7 at each of a set of different times.

圖11顯示一原始之左耳立體聲濾波器脈衝響應及一根據本發明的具體實施例之左耳立體聲濾波器脈衝響應。 Figure 11 shows an original left ear stereo filter impulse response and a left ear stereo filter impulse response in accordance with an embodiment of the present invention.

圖12顯示一原始之立體聲化和濾波器脈衝響應及一根據本發明的具體實施例之立體聲化和濾波器脈衝響應。 Figure 12 shows an original stereo and filter impulse response and a stereo and filter impulse response in accordance with an embodiment of the present invention.

圖13顯示一原始之立體聲化差濾波器脈衝響應及一根據本發明的具體實施例之立體聲化差濾波器脈衝響應。 Figure 13 shows an original stereo differential filter impulse response and a stereo differential filter impulse response in accordance with an embodiment of the present invention.

圖14A-14E顯示當作該和及差濾波器響應中之頻率的函數之能量遍及變化的時間間隔之繪圖,並沿著本發明的一對示範立體聲濾波器具體實施例之濾波器脈衝響應的長度。 Figures 14A-14E show plots of energy spread over time as a function of frequency in the sum and difference filter response, and along the filter impulse response of a pair of exemplary stereo filter embodiments of the present invention. length.

圖15A及15B顯示分別用於本發明的一對示範立體聲濾波器具體實施例之和及頻率濾波器脈衝響應的時頻平面上之相等衰減輪廓。 Figures 15A and 15B show equal attenuation profiles on the time-frequency plane of the sum of a pair of exemplary stereo filter embodiments and frequency filter impulse responses, respectively, for use with the present invention.

圖16A及16B顯示分別用於本發明的一對示範立體聲濾波器具體實施例之和及頻率濾波器脈衝響應的時頻繪 圖、亦即頻譜圖之表面的等角視圖。 16A and 16B show the time-frequency plot of the sum of a pair of exemplary stereo filter embodiments and the frequency filter impulse response for the present invention, respectively. Figure, is an isometric view of the surface of the spectrogram.

圖17A及17B顯示與圖16A及16B相同的時頻繪圖之表面的等角視圖,但分別用於一對典型立體聲濾波器之和及頻率濾波器脈衝響應,該對立體聲濾波器特別是用於待匹配之圖16A及16B的立體聲濾波器。 17A and 17B show isometric views of the same time-frequency plotting surface as in Figs. 16A and 16B, but for a sum of a pair of typical stereo filters and a frequency filter impulse response, respectively, which are used in particular for The stereo filters of Figures 16A and 16B to be matched.

圖18顯示一音頻處理設備之實施的形式,該音頻處理設備被組構成根據本發明之態樣處理一組音頻輸入信號。 Figure 18 shows a form of implementation of an audio processing device that is organized to process a set of audio input signals in accordance with aspects of the present invention.

圖19A顯示一接收五聲道音頻資訊的立體聲化設備之具體實施例的簡化方塊圖。 Figure 19A shows a simplified block diagram of a particular embodiment of a stereo device that receives five channels of audio information.

圖19B顯示一接收四聲道音頻資訊的立體聲化設備之具體實施例的簡化方塊圖。 Figure 19B shows a simplified block diagram of a particular embodiment of a stereo device that receives four-channel audio information.

603‧‧‧混洗器 603‧‧‧Breaker

605‧‧‧和濾波器 605‧‧‧ and filter

607‧‧‧差時變濾波器 607‧‧‧Differential time-varying filter

609‧‧‧解混洗器 609‧‧‧Unmixer

Claims (41)

  1. 一種用於立體聲化一組一或多個音頻輸入信號之設備,包括:一對立體聲濾波器,其以一或多對基本立體聲濾波器為特徵,一對基本立體聲濾波器用於該等音頻信號輸入之每一個,每一對基本立體聲濾波器能被一基本左耳濾波器及一基本右耳濾波器所代表,且進一步能以一基本和濾波器及一基本差濾波器所代表,每一濾波器能以一個別之脈衝響應為其特徵,其中基本立體聲濾波器之至少一對被組構成空間化其個別之音頻信號輸入,以合併從個別之虛擬喇叭位置至一收聽者的一直接響應,及合併一收聽室之早期回音與回響的響應,及其中針對該基本立體聲濾波器之至少一對:該基本和濾波器之時頻特徵實質上係與該基本差濾波器之時頻特徵不同,使得在所有頻率,該基本和濾波器長度顯著地小於該基本差濾波器長度、該基本左耳濾波器長度、及該基本右耳濾波器長度;及與該基本左耳濾波器長度或該基本右耳濾波器長度隨頻率的變化作比較,該基本和濾波器長度橫越不同頻率顯著地變化,使該基本和濾波器長度隨著增加之頻率而減少,使得該設備產生可經過耳機或在單音混合之後單音地播放的輸出信號。 An apparatus for stereophoning a set of one or more audio input signals, comprising: a pair of stereo filters characterized by one or more pairs of basic stereo filters, a pair of basic stereo filters being used for the audio signal inputs Each of the pair of basic stereo filters can be represented by a basic left ear filter and a basic right ear filter, and further can be represented by a basic sum filter and a basic difference filter, each filter The device can be characterized by a different impulse response, wherein at least one pair of the basic stereo filters are grouped to spatialize their individual audio signal inputs to combine a direct response from the individual virtual horn position to a listener, And combining the early echo and reverberation responses of a listening room, and at least one pair of the basic stereo filters: the time-frequency characteristics of the base and the filter are substantially different from the time-frequency characteristics of the basic difference filter, So that at all frequencies, the base and filter lengths are significantly smaller than the base difference filter length, the base left ear filter length, and the base The length of the right ear filter; and comparing the length of the basic left ear filter or the length of the basic right ear filter as a function of frequency, the fundamental and filter lengths vary significantly across different frequencies, such that the basic and filter The length decreases with increasing frequency, causing the device to produce an output signal that can be played monophonically through the headphones or after mixing the tones.
  2. 如申請專利範圍第1項之設備,其中針對該基本立體聲濾波器之至少一對,在該基本和濾波器脈衝響應之最初時間間隔期間,該基本和濾波器脈衝響應之變遷至一不足道位準隨著時間之消逝以頻率相依之方式逐漸地發生。 The apparatus of claim 1, wherein for at least one pair of the basic stereo filters, the basic and filter impulse responses transition to an inactive level during the initial time interval of the basic and filter impulse responses As time passes, it gradually occurs in a frequency-dependent manner.
  3. 如申請專利範圍第2項之設備,其中針對該基本立體聲濾波器之至少一對,該基本和濾波器在該變遷時間間隔期間在頻率成分中由最初全帶寬減少朝向一低頻截止。 The apparatus of claim 2, wherein for at least one pair of the basic stereo filters, the base and filter are turned off in the frequency component from the initial full bandwidth reduction toward a low frequency during the transition time interval.
  4. 如申請專利範圍第2項之設備,其中針對該基本立體聲濾波器之至少一對,該變遷時間間隔係使得該基本和濾波器脈衝響應由在最高大約3ms(毫秒)的全帶寬變遷至在大約40ms的低於100Hz(赫茲)。 The apparatus of claim 2, wherein for at least one pair of the basic stereo filters, the transition time interval is such that the basic and filter impulse response is shifted from a full bandwidth of up to about 3 ms (milliseconds) to about 40ms below 100Hz (Hz).
  5. 如申請專利範圍第1項之設備,其中針對該基本立體聲濾波器之至少一對,在高於10kHz(千赫)之高頻的基本差濾波器長度係少於40ms,在3kHz及4kHz間之頻率的基本差濾波器長度係少於100ms,且在少於2kHz之頻率,該基本差濾波器長度係少於160ms。 The apparatus of claim 1, wherein for at least one pair of the basic stereo filters, the fundamental difference filter length at a high frequency higher than 10 kHz (kilohertz) is less than 40 ms, between 3 kHz and 4 kHz. The fundamental difference filter length of the frequency is less than 100 ms, and at a frequency less than 2 kHz, the basic difference filter length is less than 160 ms.
  6. 如申請專利範圍第1項之設備,其中針對該基本立體聲濾波器之至少一對,在高於10kHz之高頻的基本差濾波器長度係少於20ms,在3kHz及4kHz間之頻率的基本差濾波器長度係少於60ms,且在少於2kHz之頻率,該基本差濾波器長度係少於120ms。 The apparatus of claim 1, wherein for at least one pair of the basic stereo filters, the fundamental difference filter length at a high frequency higher than 10 kHz is less than 20 ms, and the difference between the frequencies between 3 kHz and 4 kHz is substantially The filter length is less than 60 ms, and at frequencies less than 2 kHz, the basic difference filter length is less than 120 ms.
  7. 如申請專利範圍第1項之設備,其中針對該基本立體聲濾波器之至少一對,在高於10kHz之高頻的基本差濾波器長度係少於10ms,在3kHz及4kHz間之頻率的基本 差濾波器長度係少於40ms,且在少於2kHz之頻率,該基本差濾波器長度係少於80ms。 The apparatus of claim 1, wherein for at least one pair of the basic stereo filters, the basic difference filter length at a high frequency higher than 10 kHz is less than 10 ms, and the frequency between 3 kHz and 4 kHz is basic. The difference filter length is less than 40 ms, and at a frequency less than 2 kHz, the basic difference filter length is less than 80 ms.
  8. 如申請專利範圍第1項之設備,其中針對該基本立體聲濾波器之至少一對,該基本差濾波器長度係少於大約800ms。 The apparatus of claim 1, wherein the basic difference filter length is less than about 800 ms for at least one pair of the basic stereo filters.
  9. 如申請專利範圍第1項之設備,其中針對該基本立體聲濾波器之至少一對,該基本差濾波器長度係少於大約400ms。 The apparatus of claim 1, wherein the basic difference filter length is less than about 400 ms for at least one pair of the basic stereo filters.
  10. 如申請專利範圍第1項之設備,其中針對該基本立體聲濾波器之至少一對,該基本差濾波器長度係少於大約200ms。 The apparatus of claim 1, wherein the basic difference filter length is less than about 200 ms for at least one pair of the basic stereo filters.
  11. 如申請專利範圍第1項之設備,其中針對該基本立體聲濾波器之至少一對,該基本和濾波器長度隨著增加之頻率而減少,對於所有少於100Hz之頻率,該基本和濾波器長度係至少40ms及最多160ms,對於所有在100Hz及1kHz間之頻率,該基本和濾波器長度係至少20ms及最多80ms,對於所有在1kHz及2kHz間之頻率,該基本和濾波器長度係至少10ms及最多20ms,且對於所有在2kHz及20kHz間之頻率,該基本和濾波器長度係至少5ms及最多20ms。 The apparatus of claim 1, wherein the base and filter lengths decrease with increasing frequency for at least one pair of the basic stereo filters, and the base and filter lengths for all frequencies less than 100 Hz For at least 40 ms and at most 160 ms, the base and filter lengths are at least 20 ms and at most 80 ms for all frequencies between 100 Hz and 1 kHz. For all frequencies between 1 kHz and 2 kHz, the basic and filter lengths are at least 10 ms and Up to 20 ms, and for all frequencies between 2 kHz and 20 kHz, the base and filter length is at least 5 ms and at most 20 ms.
  12. 如申請專利範圍第1項之設備,其中針對該基本立體聲濾波器之至少一對, 該基本和濾波器長度隨著增加之頻率而減少,對於所有少於100Hz之頻率,該基本和濾波器長度係至少60ms及最多120ms,對於所有在100Hz及1kHz間之頻率,該基本和濾波器長度係至少30ms及最多60ms,對於所有在1kHz及2kHz間之頻率,該基本和濾波器長度係至少15ms及最多30ms,且對於所有在2kHz及20kHz間之頻率,該基本和濾波器長度係至少7ms及最多15ms。 An apparatus as claimed in claim 1, wherein at least one pair of the basic stereo filters is The base and filter lengths decrease with increasing frequency. For all frequencies less than 100 Hz, the base and filter lengths are at least 60 ms and at most 120 ms. For all frequencies between 100 Hz and 1 kHz, the base and filter The length is at least 30 ms and at most 60 ms. For all frequencies between 1 kHz and 2 kHz, the basic and filter lengths are at least 15 ms and at most 30 ms, and for all frequencies between 2 kHz and 20 kHz, the basic and filter lengths are at least 7ms and up to 15ms.
  13. 如申請專利範圍第1項之設備,其中針對該基本立體聲濾波器之至少一對,該基本和濾波器長度隨著增加之頻率而減少,對於所有少於100Hz之頻率,該基本和濾波器長度係至少70ms及最多90ms,對於所有在100Hz及1kHz間之頻率,該基本和濾波器長度係至少35ms及最多50ms,對於所有在1kHz及2kHz間之頻率,該基本和濾波器長度係至少18ms及最多25ms,且對於所有在2kHz及20kHz間之頻率,該基本和濾波器長度係至少8ms及最多12ms。 The apparatus of claim 1, wherein the base and filter lengths decrease with increasing frequency for at least one pair of the basic stereo filters, and the base and filter lengths for all frequencies less than 100 Hz For at least 70 ms and at most 90 ms, the base and filter lengths are at least 35 ms and at most 50 ms for all frequencies between 100 Hz and 1 kHz. For all frequencies between 1 kHz and 2 kHz, the basic and filter lengths are at least 18 ms and Up to 25ms, and for all frequencies between 2kHz and 20kHz, the base and filter length is at least 8ms and up to 12ms.
  14. 如申請專利範圍第1至13項的任一項之設備,其中針對該基本立體聲濾波器之至少一對,該等基本立體聲濾波器特徵係由一對待匹配立體聲濾波器特徵所決定。 The apparatus of any one of claims 1 to 13, wherein for at least one pair of the basic stereo filters, the basic stereo filter characteristics are determined by a feature to be matched to the stereo filter.
  15. 如申請專利範圍第14項之設備,其中針對該基本 立體聲濾波器之至少一對,該基本差濾波器脈衝響應係在晚些時候實質上與該待匹配立體聲濾波器之差濾波器成比例。 Such as the equipment of claim 14 of the patent scope, wherein the basic At least one pair of stereo filters, the substantially differential filter impulse response is substantially proportional to the difference filter of the stereo filter to be matched at a later time.
  16. 如申請專利範圍第15項之設備,其中針對該基本立體聲濾波器之至少一對,該基本差濾波器脈衝響應在40ms之後變得實質上與該待匹配立體聲濾波器之差濾波器成比例。 The apparatus of claim 15 wherein, for at least one pair of the basic stereo filters, the basic difference filter impulse response becomes substantially proportional to a difference filter of the stereo filter to be matched after 40 ms.
  17. 一種用於立體聲化一組一或多個音頻輸入信號之方法,該方法包括:藉由立體聲化器(binauralizer)過濾該組音頻輸入信號,該立體聲化器以一或多對基本立體聲濾波器為其特徵,一對基本立體聲濾波器用於該等音頻信號輸入之每一個,每一對基本立體聲濾波器能被一基本左耳濾波器及一基本右耳濾波器所代表,且進一步能以一基本和濾波器及一基本差濾波器所代表,每一濾波器能以一個別之脈衝響應為其特徵,其中基本立體聲濾波器之至少一對被組構成空間化其個別之音頻信號輸入,以合併從個別之虛擬喇叭位置至一收聽者的一直接響應,及合併一收聽室之早期回音與回響的響應,及其中針對該基本立體聲濾波器之至少一對:該基本和濾波器之時頻特徵實質上係與該基本差濾波器之時頻特徵不同,使得在所有頻率,該基本和濾波器長度比該基本差濾波器長度、該基本左耳濾波器長度、 及該基本右耳濾波器長度顯著地較小;及與該基本左耳濾波器長度或該基本右耳濾波器長度隨頻率的變化作比較,該基本和濾波器長度橫越不同頻率顯著地變化,使該基本和濾波器長度隨著增加之頻率而減少,使得該等輸出係可經過耳機或單音地播放。 A method for stereophoning a set of one or more audio input signals, the method comprising: filtering the set of audio input signals by a binauralizer, the stereoizer being in one or more pairs of basic stereo filters Characterized by a pair of basic stereo filters for each of the audio signal inputs, each pair of basic stereo filters can be represented by a basic left ear filter and a basic right ear filter, and further capable of And a filter and a basic difference filter, each filter being characterized by a different impulse response, wherein at least one pair of the basic stereo filters are grouped to spatialize their individual audio signal inputs for merging a direct response from an individual virtual horn position to a listener, and a response to the early echo and reverberation of a listening room, and at least one pair of the basic stereo filters: time-frequency characteristics of the base and filter Substantially different from the time-frequency characteristic of the basic difference filter, such that the fundamental and filter lengths are more than the basic difference filter at all frequencies Length, the base left ear filter length, And the length of the basic right ear filter is significantly smaller; and compared to the change in the length of the basic left ear filter or the length of the basic right ear filter, the fundamental and filter lengths vary significantly across different frequencies The base and filter lengths are reduced with increasing frequency so that the output can be played through the headphones or in a single tone.
  18. 如申請專利範圍第17項之方法,其中針對該基本立體聲濾波器之至少一對,在該基本和濾波器脈衝響應之最初時間間隔期間,該基本和濾波器脈衝響應之變遷至一不足道位準隨著時間之消逝以頻率相依之方式逐漸地發生。 The method of claim 17, wherein for at least one pair of the basic stereo filters, the basic and filter impulse responses transition to an inactive level during the initial time interval of the basic and filter impulse responses As time passes, it gradually occurs in a frequency-dependent manner.
  19. 如申請專利範圍第18項之方法,其中針對該基本立體聲濾波器之至少一對,該基本和濾波器在該變遷時間間隔期間在頻率成分中由最初全帶寬減少朝向一低頻截止。 The method of claim 18, wherein for at least one pair of the basic stereo filters, the base and filter are turned off in the frequency component from the initial full bandwidth reduction toward a low frequency during the transition time interval.
  20. 如申請專利範圍第18項之方法,其中針對該基本立體聲濾波器之至少一對,該變遷時間間隔係使得該基本和濾波器脈衝響應由在最高大約30ms的全帶寬變遷至在大約40ms的低於100Hz。 The method of claim 18, wherein for at least one pair of the basic stereo filters, the transition time interval is such that the basic and filter impulse response is shifted from a full bandwidth of up to about 30 ms to a low of about 40 ms. At 100Hz.
  21. 如申請專利範圍第17至20項的任一項之方法,其中針對該基本立體聲濾波器之至少一對,在高於10kHz之高頻的基本差濾波器長度係少於40ms,在3kHz及4kHz間之頻率的基本差濾波器長度係少於100ms,且在少於2kHz之頻率,該基本差濾波器長度係少於160ms。 The method of any one of clauses 17 to 20, wherein for at least one pair of the basic stereo filters, the fundamental difference filter length at a high frequency above 10 kHz is less than 40 ms, at 3 kHz and 4 kHz. The fundamental difference filter length of the frequency is less than 100 ms, and at a frequency less than 2 kHz, the basic difference filter length is less than 160 ms.
  22. 如申請專利範圍第17項之方法,其中針對該基本立體聲濾波器之至少一對,在高於10kHz之高頻的基本差濾波器長度係少於20ms,在3kHz及4kHz間之頻率的基本差濾波器長度係少於60ms,且在少於2kHz之頻率,該基本差濾波器長度係少於120ms。 The method of claim 17, wherein for at least one pair of the basic stereo filters, the fundamental difference filter length at a high frequency higher than 10 kHz is less than 20 ms, and the difference between the frequencies between 3 kHz and 4 kHz is substantially The filter length is less than 60 ms, and at frequencies less than 2 kHz, the basic difference filter length is less than 120 ms.
  23. 如申請專利範圍第17項之方法,其中針對該基本立體聲濾波器之至少一對,在高於10kHz之高頻的基本差濾波器長度係少於10ms,在3kHz及4kHz間之頻率的基本差濾波器長度係少於40ms,且在少於2kHz之頻率,該基本差濾波器長度係少於80ms。 The method of claim 17, wherein for at least one pair of the basic stereo filters, the fundamental difference filter length at a high frequency higher than 10 kHz is less than 10 ms, and the difference between the frequencies between 3 kHz and 4 kHz is substantially The filter length is less than 40 ms, and at frequencies less than 2 kHz, the basic difference filter length is less than 80 ms.
  24. 如申請專利範圍第17項之方法,其中針對該基本立體聲濾波器之至少一對,該基本差濾波器長度係少於大約800ms。 The method of claim 17, wherein the basic difference filter length is less than about 800 ms for at least one pair of the basic stereo filters.
  25. 如申請專利範圍第17項之方法,其中針對該基本立體聲濾波器之至少一對,該基本差濾波器長度係少於大約400ms。 The method of claim 17, wherein the basic difference filter length is less than about 400 ms for at least one pair of the basic stereo filters.
  26. 如申請專利範圍第17項之方法,其中針對該基本立體聲濾波器之至少一對,該基本差濾波器長度係少於大約200ms。 The method of claim 17, wherein the basic difference filter length is less than about 200 ms for at least one pair of the basic stereo filters.
  27. 如申請專利範圍第17項之方法,其中針對該基本立體聲濾波器之至少一對,該基本和濾波器長度隨著增加之頻率而減少,對於所有少於100Hz之頻率,該基本和濾波器長度係至少40ms及最多160ms, 對於所有在100Hz及1kHz間之頻率,該基本和濾波器長度係至少20ms及最多80ms,對於所有在1kHz及2kHz間之頻率,該基本和濾波器長度係至少10ms及最多20ms,且對於所有在2kHz及20kHz間之頻率,該基本和濾波器長度係至少5ms及最多20ms。 The method of claim 17, wherein the base and filter lengths decrease with increasing frequency for at least one pair of the basic stereo filters, the base and filter length for all frequencies less than 100 Hz At least 40ms and up to 160ms, For all frequencies between 100 Hz and 1 kHz, the basic and filter lengths are at least 20 ms and at most 80 ms. For all frequencies between 1 kHz and 2 kHz, the basic and filter lengths are at least 10 ms and at most 20 ms, and for all The frequency between 2 kHz and 20 kHz is at least 5 ms and at most 20 ms.
  28. 如申請專利範圍第17項之方法,其中針對該基本立體聲濾波器之至少一對,該基本和濾波器長度隨著增加之頻率而減少,對於所有少於100Hz之頻率,該基本和濾波器長度係至少60ms及最多120ms,對於所有在100Hz及1kHz間之頻率,該基本和濾波器長度係至少30ms及最多60ms,對於所有在1kHz及2kHz間之頻率,該基本和濾波器長度係至少15ms及最多30ms,且對於所有在2kHz及20kHz間之頻率,該基本和濾波器長度係至少7ms及最多15ms。 The method of claim 17, wherein the base and filter lengths decrease with increasing frequency for at least one pair of the basic stereo filters, the base and filter length for all frequencies less than 100 Hz For at least 60 ms and at most 120 ms, the base and filter lengths are at least 30 ms and at most 60 ms for all frequencies between 100 Hz and 1 kHz. For all frequencies between 1 kHz and 2 kHz, the basic and filter lengths are at least 15 ms and Up to 30 ms, and for all frequencies between 2 kHz and 20 kHz, the base and filter length is at least 7 ms and at most 15 ms.
  29. 如申請專利範圍第17項之方法,其中針對該基本立體聲濾波器之至少一對,該基本和濾波器長度隨著增加之頻率而減少,對於所有少於100Hz之頻率,該基本和濾波器長度係至少70ms及最多90ms,對於所有在100Hz及1kHz間之頻率,該基本和濾波器長度係至少35ms及最多50ms, 對於所有在1kHz及2kHz間之頻率,該基本和濾波器長度係至少18ms及最多25ms,且對於所有在2kHz及20kHz間之頻率,該基本和濾波器長度係至少8ms及最多12ms。 The method of claim 17, wherein the base and filter lengths decrease with increasing frequency for at least one pair of the basic stereo filters, the base and filter length for all frequencies less than 100 Hz For at least 70 ms and up to 90 ms, the base and filter lengths are at least 35 ms and at most 50 ms for all frequencies between 100 Hz and 1 kHz. The base and filter lengths are at least 18 ms and at most 25 ms for all frequencies between 1 kHz and 2 kHz, and for all frequencies between 2 kHz and 20 kHz, the base and filter lengths are at least 8 ms and at most 12 ms.
  30. 如申請專利範圍第17至20項的任一項之方法,其中針對該基本立體聲濾波器之至少一對,該等基本立體聲濾波器特徵係由一對待匹配立體聲濾波器特徵所決定。 The method of any one of clauses 17 to 20, wherein the at least one pair of the basic stereo filters are determined by a characteristic of the stereo filter to be matched.
  31. 一種用於立體聲化的操作信號處理設備之方法,該方法包括:接收一對信號,該等信號代表被組構成立體聲化一音頻信號的對應待匹配立體聲濾波器對之脈衝響應;藉由一對濾波器處理該對被接收之信號,每一濾波器係以具有時變濾波器特徵之修改濾波器為其特徵,該處理形成一對代表對應之被修改立體聲濾波器對的脈衝響應之被修改信號,使得該等被修改之立體聲濾波器被組構成立體聲化一音頻信號,且另具有單音下降混合中之低感知回響、與遍及耳機的對立體聲濾波器之最小衝擊的特性。 A method for operating a signal processing apparatus for stereophony, the method comprising: receiving a pair of signals representing impulse responses of pairs of stereo signals to be matched that are stereoscopically formed into an audio signal; A filter processes the pair of received signals, each filter being characterized by a modified filter having a time varying filter characteristic that forms a pair of modified impulse responses representing pairs of modified stereo filters. The signals are such that the modified stereo filters are grouped to form a stereo-audio signal, and additionally have a low perceptual reverberation in monophonic downmixing and a minimum impact on the stereo filter across the headphones.
  32. 如申請專利範圍第31項之方法,其中被修改之立體聲濾波器係以一被修改之和濾波器及一被修改之差濾波器為其特徵,且其中該時變濾波器被組構成使得:被修改之立體聲濾波器脈衝響應包括一藉由頭部相關轉移函數所界定之直接部份,用於收聽者在一預先確定位置收聽一虛擬之喇叭; 與該被修改之差濾波器作比較,該被修改之和濾波器具有一顯著地減少之位準及一顯著地較短之回響時間,及由該和濾波器之脈衝響應的直接部份至該和濾波器之可忽略的響應部份有一平順之變遷,使平順之變遷具隨著時間之消逝的頻率選擇性。 The method of claim 31, wherein the modified stereo filter is characterized by a modified sum filter and a modified difference filter, and wherein the time varying filter is grouped such that: The modified stereo filter impulse response includes a direct portion defined by a head related transfer function for the listener to listen to a virtual horn at a predetermined position; Comparing to the modified difference filter, the modified sum filter has a significantly reduced level and a significantly shorter reverberation time, and the direct portion of the impulse response of the sum filter And the negligible response portion of the filter has a smooth transition that makes the smooth transition with frequency selectivity that fades over time.
  33. 一種用於立體聲化的操作信號處理設備之方法,該方法包括:接收代表對應於左耳及右耳立體聲濾波器之脈衝響應的左耳信號及右耳信號,該等立體聲濾波器被組構成立體聲化一音頻信號;混洗該左耳信號及右耳信號,以形成一與該左及右耳信號之和成比例的和信號、及一與該左耳信號及該右耳信號間之差成比例的差信號;藉由一具有時變濾波器特徵之和濾波器過濾該和信號,該過濾形成一被過濾之和信號;藉由一以該和濾波器為其特徵之差濾波器處理該差信號,該處理形成一被過濾之差信號;解混洗該被過濾之和信號及該被過濾之差信號,以形成代表對應於左耳及右耳被修改的立體聲濾波器之脈衝響應的被修改之左耳信號及被修改之右耳信號,其中該等被修改之立體聲濾波器被組構成立體聲化一音頻信號,可藉由一被修改之和濾波器及一被修改之差濾波器所代表,且另具有如申請專利範圍第1項所陳述之用於立體聲化一組一或多個音頻輸入信號之設備。 A method for operating a signal processing apparatus for stereophony, the method comprising: receiving a left ear signal and a right ear signal representing impulse responses corresponding to left and right ear stereo filters, the stereo filters being grouped to form a stereo Forming an audio signal; shuffling the left ear signal and the right ear signal to form a sum signal proportional to the sum of the left and right ear signals, and a difference between the left ear signal and the right ear signal a proportional difference signal; filtering the sum signal by a sum filter having a time varying filter characteristic, the filtering forming a filtered sum signal; processing the difference by a difference filter characterized by the sum filter a difference signal, the process forming a filtered difference signal; deshuffling the filtered sum signal and the filtered difference signal to form an impulse response representative of a stereo filter corresponding to the modified left and right ears The modified left ear signal and the modified right ear signal, wherein the modified stereo filters are grouped to form a stereo audio signal, which can be modified by a modified sum filter and a modified It represents the difference filter, and the other patent as having a range of item set forth a device or a plurality of audio input signals for stereophonic sound.
  34. 如申請專利範圍第33項之方法,其中該被修改之和信號被適當地提升,以補償該時變過濾所造成之被修改的差信號中之任何失去的能量。 The method of claim 33, wherein the modified sum signal is appropriately boosted to compensate for any lost energy in the modified difference signal caused by the time varying filtering.
  35. 如申請專利範圍第31項之方法,其中修改之該時變濾波器係能以在一代表該等待匹配立體聲濾波器之和濾波器的信號上操作之和修改濾波器、及在一代表該等待匹配立體聲濾波器之差濾波器的信號上操作之差修改濾波器所代表,其中針對比40ms稍後之時間,該和修改濾波器實質上衰減代表該等待匹配立體聲濾波器之和濾波器的信號,及其中該差修改濾波器係可藉由該和修改濾波器之時變特徵所界定。 The method of claim 31, wherein the modified time varying filter is capable of modifying the filter with a sum of operations on a signal representative of the sum filter matching the stereo filter, and The difference between the operationally modified filters of the difference filter matching the stereo filter is represented by the filter, wherein the modified filter substantially attenuates the signal representative of the sum filter of the wait-matched stereo filter for a later time than 40 ms. And the difference modifying filter therein can be defined by the time varying characteristics of the summing filter.
  36. 如申請專利範圍第35項之方法,其中該和修改濾波器係能以在標示為t之時間對在時間t=τ的脈衝之時變脈衝響應f(t,τ)為其特徵,且其中該和修改濾波器係亦能以包括一時變帶寬之時變頻率響應為其特徵,其中該差修改濾波器之脈衝響應係可由f(t,τ)決定,且其中該時變帶寬係及時單調遞減。 The method of claim 35, wherein the modified filter system is characterized by a time-varying impulse response f(t, τ ) of a pulse at time t= τ at a time labeled t, and wherein The modified filter system can also be characterized by a time varying frequency response including a time varying bandwidth, wherein the impulse response of the differential modified filter can be determined by f(t, τ ), and wherein the time varying bandwidth is monotonous in time Decrement.
  37. 如申請專利範圍第36項之方法,其中用在大於大約40ms之時間,該時變帶寬平順地減少至少於100Hz。 The method of claim 36, wherein the time varying bandwidth is reduced by at least 100 Hz in a time greater than about 40 ms.
  38. 如申請專利範圍第36至37項的任一項之方法,其中該差修改濾波器之脈衝響應係與成比例,其中h D0(t)標示從該混洗所得之該差信號。 The method of any one of claims 36 to 37, wherein the differentially modified filter has an impulse response system Proportional, where h D 0 ( t ) indicates the difference signal obtained from the shuffling.
  39. 一種用於立體聲化的程式邏輯,其當藉由一處理系統之至少一處理器所執行時,造成實行一如申請專利範圍第17項之方法。 A program logic for stereos that, when executed by at least one processor of a processing system, results in a method as in claim 17 of the patent application.
  40. 一種用於立體聲化的電腦可讀取媒體,在其中具有程式邏輯,當藉由一處理系統之至少一處理器執行該程式邏輯時,造成實行一如申請專利範圍第17項之方法。 A computer readable medium for stereo, having program logic therein, which when executed by at least one processor of a processing system, results in a method as claimed in claim 17.
  41. 一種用於立體聲化的設備,包括:一處理系統,其包括:至少一處理器,及一儲存裝置,其中該儲存裝置被組構成具有程式邏輯,當執行該程式邏輯時,造成該設備實行如申請專利範圍第17項之方法。 A device for stereoscopicization, comprising: a processing system comprising: at least one processor, and a storage device, wherein the storage device is grouped to have program logic, when the program logic is executed, causing the device to perform The method of applying for the scope of patent item 17.
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