TW479429B - System and method for distributing telephone audio data via a computer network - Google Patents

System and method for distributing telephone audio data via a computer network Download PDF

Info

Publication number
TW479429B
TW479429B TW89119345A TW89119345A TW479429B TW 479429 B TW479429 B TW 479429B TW 89119345 A TW89119345 A TW 89119345A TW 89119345 A TW89119345 A TW 89119345A TW 479429 B TW479429 B TW 479429B
Authority
TW
Taiwan
Prior art keywords
data
voice data
voice
computer
item
Prior art date
Application number
TW89119345A
Other languages
Chinese (zh)
Inventor
Lynda J Meyer
Jeffrey G Markel
Jeffrey O'connell
Original Assignee
Net Technologies Inc
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Net Technologies Inc filed Critical Net Technologies Inc
Application granted granted Critical
Publication of TW479429B publication Critical patent/TW479429B/en

Links

Abstract

A method for streaming telephone audio data over a computer network, the method comprising: receiving the audio data via a telephone call from an author of the audio data; directly encoding the audio data into a destination file format; and storing the encoded audio data on a file system accessible by a computer.

Description

^+/^429 ^+/^429 五、發明說明d) 申請人特此 臨時申請案以 申請案之利益 本專利文件 所有人對於由 並無異議,就 的一樣,但是 在此 發明尤 有關, 網際 要的通 媒體, 業有很 但是在 版他們 他.議題 在廣 而不管 文字作 第三方 視廣播 公佈的 其與接 該電話 網路已 訊成就 可讓他 南的入 網際網 的作品 0 播界内 允許他 者相同 所設下 散佈他 申請1999年9月20日歸檔的第60/154, 769號 及2000年2月2日歸檔的第60/179, 831號臨時 ’在此將此兩案併入當成本申請案的參考。 版權注意事項 的公開部分有牽涉到版權保護的議題。版權 任何專利文件或專利公佈之人進行複本生產 如同在專利與商標局專利檔案或紀錄中出現 不論如何在其他方面則保留所有版權權利。 發明背景 :二通常來說與語音資料流系統有關,本 路的流式語音資料之系統與方法 ,.罔路透過電腦網路到達用戶端 經^告了自從製版印刷出現之後可能是最重 ,網際網路提供世界各地 · 們公佈作品、進行教·^ ^ _們—個全新的 門門檻,包含編:::t ,統上出版 路興起之:2司所在地的需求。 <後,作者可以任直 ,讓他們解決& & # /、滿思的方式出 决出版商不可能接受的爭議或其 ,類似的限制也加諸在音 們呈現的公佈領域。音訊:視::訊作者上, 的方式使用_ π _ # 與視吼作者可以與 的限制。_此個籀/、作扣,不用受到 們的訊自,;二再利用公眾無線電/電 一而疋利用公眾網際網4;藉此可^ + / ^ 429 ^ + / ^ 429 V. Description of the invention d) The applicant hereby submits the provisional application for the benefit of the application. The owner of this patent document has no objection to the cause, the same, but the invention is particularly relevant here. The Internet There is a lot of communication media, but they are in print. The issue is in the broadcast, regardless of the text, as a third-party broadcast, and its connection with the telephone network has been achieved. It can allow him to enter the Internet. The other allows the same to be used to distribute his application No. 60/154, 769 filed on September 20, 1999 and No. 60/179, 831 filed on February 2, 2000. Here are two cases Incorporated as a reference in the current cost application. The public part of the copyright notice has issues related to copyright protection. Copyright Copy production of any patent document or patent issuance, as it appears in the Patent and Trademark Office patent file or records, and in all other respects all copyright rights are reserved. Background of the invention: Secondly, it is generally related to the voice data stream system. The system and method of the stream voice data of this road. Kushiro reaches the client through a computer network. It has been reported that it may be the heaviest since the plate printing appeared The Internet provides us all around the world to publish their works and teach them ^ ^ _ _ — a brand new threshold, including editors :: t, unified publishing road rises: the needs of the 2 divisions. After that, the authors can let them settle directly and let them resolve the issue in a way that the publisher cannot accept disputes or similar problems. Similar restrictions are also imposed on the field of publication presented by the audio. Audio: The author of the video ::, uses the _ π _ # and the restriction of the author of the roar. _This deduction / deduction does not need to be received from us; second, the use of public radio / electricity, and at the same time the use of the public Internet 4;

479429 五、發明說明(2) 省略掉傳統廣 門限制後,像 可延展性與無 司都有能力透 形式之語音給 技術結合所呈 電腦有許多 傳遞,但是存 無法接收的語 自電話網路語479429 V. Description of the invention (2) After omitting the traditional wide-ranging restrictions, such as the scalability and the ability to communicate through the form of speech to the technology, the computer has many transmissions, but there are unreceivable languages.

Boxπ, 與電話 的資料 話網路 散佈。 且因一 其他 的問題 放入預 取語音 者同時 些資料 公司 拉多州 這是一 網路之 ,而另 並且透 不過此 次只能 解決方 。當接 定目錄 資料並 紀錄語 。至多 废商對 科羅拉 播基礎設施以及其諸多限制。在消除這些入 是個人電子報這類的應用就可成真。當這種 所不在的公小電話網路結合之後,個人與公 過全球共同使用的簡單電話設備,傳送^何 本質上不同的電腦用戶。的確,電腦與語音 現出來的應用是無遠弗屆的。 不同方式接收語音資料以透過電腦網路重新 在著系統無法提供即時和所需廣播或提供了 音品質之問題。有一種可讓電腦系統接收來 音資料的方法,就是使用所謂的” Gentner 種複合語音結合器,這個盒子用來當成電腦 間的介面。盒子上的一個輸入收發來自電腦 個輸入則連接至電廷網路。資料接收自電 過Gentner Box傳遞給電腦,進行編碼以及 解決方案牽涉到一次只能擁有一個連線,並 編碼一個作業的缺點。 案則有同時紀錄複數個來自電話網路資料流 收到語音資料,系統會將它儲存成檔案並且 中。一般來說,透過使用UNIX cr〇n j〇b讀 且傳送給編碼器。雖然此設定可讓多位通^舌 音資料’但卻無法即時透過電腦網路傳播這 ,該系統僅適合需求式的語音資料傳播。 於這個問題提供有解決方案,例如位於科羅 多泉的Tel ISoft Inc·就研發出一種系統,'Boxπ, data with telephone, Internet distribution. And because of other problems put into the prefetcher and some data companies. Lado State This is a network, and it can't be resolved at this time. When accessing directory information and epilogues. At best, scrap merchants impose restrictions on Corolla's infrastructure and its many restrictions. Applications such as personal newsletters can be eliminated in eliminating these entries. When this ubiquitous network of public and small telephones is combined, individuals and public telephones are used by simple telephone equipment that are used by the world to transmit essentially different computer users. Indeed, computer and voice applications are far-reaching. Receiving voice data in different ways to reconnect over the computer network. The system is unable to provide real-time and required broadcasts or provides audio quality issues. One method that allows computer systems to receive incoming audio data is to use the so-called "Gentner compound speech coupler. This box is used as an interface between computers. One input on the box is sent and received from the computer and one input is connected to the electronic court. Network. Data received from the Gentner Box is transmitted to the computer, coding and solutions involve the disadvantages of having only one connection at a time and coding one operation. The case has recorded multiple data streams from the telephone network at the same time. To the voice data, the system will save it as a file and in general. Generally speaking, it is read and sent to the encoder by using UNIX cr0nj〇b. Although this setting allows multiple users to communicate with each other, it is not real-time. This is transmitted through a computer network, and the system is only suitable for on-demand voice data transmission. There is a solution to this problem. For example, Tel ISoft Inc., located in Corodo Spring, has developed a system, '

479429 五、發明說明(3) 用於編碼與傳播接收自電話網路的語音資料。丁 e 1 1 S 〇 f t對 該問題提供「一么」解決方案,使用許多必須合作以執行 所需功能的服務。此絕緣缺乏將會限制允許系統以最佳品 質接收與編碼多重語音訊號的延展性。因此,當記錄時語 音品質是一項重要考量時,該解決方案就不適合。 第5, 675,507(‘507)號’標題為”Message storage and479429 V. Description of the invention (3) It is used to encode and disseminate the voice data received from the telephone network. D e 1 1 S 〇 f t provides a "one" solution to this problem, using many services that must cooperate to perform the required function. This lack of insulation will limit the scalability that allows the system to receive and encode multiple voice signals with optimal quality. Therefore, when the quality of the voice is an important consideration when recording, this solution is not suitable. No. 5,675,507 (‘507)’ is entitled “Message storage and

Delivery System”的美國專利提供另一種此問題的解決方 案。‘ 5 0 7專利討論一種接收及擴散傳真、語音與資料訊息 的系統及方法,該系統擷取儲存於ν〇χ或AD/pCM格式内的 =,,這些格式是一種壓縮過的pCM資料型式。然後此壓 縮貪料將根據使用者的喜好轉換成AU *WAV格式。‘ 5〇7專 ,藉由使用VOX或AD/PCM這類中介檔案當成轉換基礎,將 提供最差的語音品質。 昭需要一種系統與方法,以一種有效率並且即時和依 ^的方法編碼與傳播電話語音資料,而其語音品質與 糸統所提供的比較起來有天壤之別。 本^ 發明簡要概述 次心^月的目的就是提供一種透過電腦網路傳播電話語 貝料的系統與方法。The "Payment System" US patent provides another solution to this problem. The '507 patent discusses a system and method for receiving and diffusing fax, voice, and data messages. The system retrieves and stores the data in νχ or AD / pCM format. In the =, these formats are a compressed pCM data format. Then the compressed data will be converted into the AU * WAV format according to the user's preferences. 507 special, by using VOX or AD / PCM, etc. The intermediary file serves as the basis for conversion and will provide the worst voice quality. A system and method is needed to encode and disseminate telephone voice data in an efficient, instant and timely manner, and the voice quality is compared with that provided by the system It looks like a big difference. The purpose of this invention is to provide a system and method for disseminating phonetic materials through a computer network.

—本t明的另一個目的是提供一種依照一組預定喜好言5 疋本:日自訂方式處置語音資料之系統與方法。 過# 1月的另一個目的是提供一種網路用戶端,以融令 〇二^ ^路傳播電話語音資料的系統與方法。 收私活浯音資料並且直接將資料編碼成目的相—Another purpose of the present invention is to provide a system and method for processing voice data according to a set of predetermined preferences. Another objective of Passing January is to provide a system and method for network clients to disseminate telephone voice data on the Internet. Receive private live sound data and directly encode the data into a purpose photo

479429 五 發明說明(4) 格式,而不需要中間語音資料轉換(即是不用 檔案格式)的系統可達成上述以及其他目的。: 理產生相當程度改善的語音品質。 星接編碼處 有一種系統與方法也可達成某些本發明的 目的,該系統與方法接受使用者用電話嗖,以及其他 並且透過電腦網路(像是網際網路) 儲路 =儲存裝置内,或兩者同時進行。當接路 乐統3 =糸統貢料庫中取得使用者的個人簡介資料。使用 者個人簡介包含系統用來決定如何處置語音資料的資訊,479429 V. Description of the invention (4) A system that does not require intermediate voice data conversion (that is, does not require a file format) can achieve the above and other objectives. : Management produces considerable improved speech quality. There is also a system and method at the star connection code that can achieve some of the objects of the present invention. The system and method accepts telephone calls from users, and other computer networks (such as the Internet). , Or both. Dangtong Road Le Tong 3 = Get the user's personal profile information in the Tong Tong Tribune. The user profile contains information that the system uses to decide what to do with the voice data.

像是是否應由特定資料流語音伺服器即時傳播以及/或在 ΐ i ί置上進行壓縮、用來壓縮資料的編碼器、壓縮過的 浯二1料檔名等等。該擷取的資料會依照xml(可延伸式標 ^σ)法則格式化’該法則定義資料必須依附以便系統 繼續執行的格式。另外,本發明也可使用其他專有的訊息 格式化法則。Such as whether it should be transmitted by a specific stream voice server in real time and / or compressed on the server, the encoder used to compress the data, the compressed file name, etc. The retrieved data will be formatted according to the xml (extendable standard ^ σ) rule ', which defines the format that the data must be attached for the system to continue to execute. In addition, the present invention can also use other proprietary message formatting rules.

e XML描述槽與語音資料將導引到語音繞送軟體,其將如 3使用者的X ML描述檔所定義般將送來之資料流傳送到適 ^的編碼器。資料在繞送到適當編碼器之後,將依照許多 可用、的代碼演算法之一進行壓縮。系統使用的範例壓縮/ 解聖細程式包含QuickTime和RealAudio。這些資料將完全 依照使用者XML描述檔中的資料進行壓縮或放置在檔案系 、洗中 由資料流语音伺服器壓縮或傳播。 5亥系統也有工作流程以及出版者軟體,這兩個子系統依 照工作流程内含的使用者id就可從系統資料庫中取得語音e XML description slot and voice data will lead to voice routing software, which will send the incoming data stream to the appropriate encoder as defined by the 3 user's X ML description file. After the data is routed to the appropriate encoder, it is compressed according to one of many available code algorithms. Sample compression / decompression programs used by the system include QuickTime and RealAudio. These data will be compressed or placed in the file system completely according to the data in the user's XML description file, compressed or transmitted by the data stream voice server. The 5H system also has a workflow and publisher software. These two subsystems can obtain voice from the system database according to the user id included in the workflow.

479429479429

五、發明說明(5) 寅料的文字位址,並且利用任何方式傳遞到任何目的地 :如;:將位置寫入電子郵件内炎且傳送給使用者。另。 ^ ’傳Λ ΓΛ文字寫成樓案並透過FTP(槽案傳送通訊協 =)專^到罔路伺服器的特定目錄内。飼服器從棺案中讀 社。:拉ΐ且可依照使用者的要求將它嵌入當成網頁連、 二料二ή i ΐ求的網頁時,該連結就會起動並且將語音 貝^、專迗到要求的地方進行播放。V. Description of the invention (5) The text address of the material and pass it to any destination by any means: such as: write the location into the email and send it to the user. another. ^ "Transfer Λ ΓΛ text into a building case and transfer it to a specific directory of Kushiro server via FTP (slot case transmission communication protocol =). The feeder was read from the coffin case. : Pull and embed it as a web page link according to the user's request. The link will be activated and the voice will be played to the requested place.

而口為接收與編碼點之間的中介檔案袼式已經不需要 二’:以可增強系統效能。從電話網路接收原始PCM資 ^斗’心後直接編碼成目的檔案袼式(像是ReaUudi〇、 = 二^或%11^0…袼式),如此就可獲得優異的 =曰^質。然後這些直接編碼的語音檔案將即時或依照要 求,透過電腦網路傳送給資料流語音用戶端。The media file format between receiving and encoding points is no longer required. It can enhance system performance. After receiving the original PCM data from the telephone network, you can directly encode it into a destination file format (such as ReaUudi 0, = 2 ^ or% 11 ^ 0 ...), so you can get excellent = ^^ quality. These directly encoded voice files will then be sent to the streaming voice client over the computer network in real time or as required.

:2由分散在不同電腦上執行的軟體間之系統負荷,藉 次祖^任何電腦上的工作負擔也可提昇效能。另外因為將 =枓私定給執行備援軟體的電腦就可輕易將它帶入處理 士所以此杈組化設計也有擴充性。例如,若大量使用者 二盼要編碼成QuickTime袼式,就可讓更多QuickTime編碼 上線。更進一步,同時將進入的語音資料流繞送給複數 個j碼器可迅速完成處理。藉由將語音檔案引導到特定伺 ,裔叢集,系統便可就地編碼,在此就可產生大量的資料 二’並將編碼過以及壓縮過的檔案散佈到網路各地的伺服 叩上。結合這些好處將可提供增強的效能以及利用程度最 佳的硬體資源。: 2 The system load is distributed among software running on different computers. The workload on any computer can also improve performance. In addition, because it is privately assigned to the computer running the backup software, it can be easily brought into the processor, so the group design is also extensible. For example, if a large number of users want to encode in QuickTime mode, more QuickTime encodings can be brought online. Furthermore, at the same time, the incoming voice data stream is routed to a plurality of j-coders at the same time, which can quickly complete the processing. By directing the voice files to a specific server and cluster, the system can encode them in situ, where a large amount of data can be generated. Second, the encoded and compressed files are distributed to the server 叩 around the network. Combining these benefits will provide enhanced performance and the best utilization of hardware resources.

479429 五、發明說明(6) 圖式之簡單說明 本發明將以附圖圖式做說明,其用途僅在範例說明並不 做限制,其中相同的參考號碼用於指示相同或對應的部 份,其中: 圖1為依照本發明具體實施例,呈現出本系統許多軟硬 體組件的設定之圖式; 圖2呈現出設定當成應用服務提供者(ASP)執行的本發明 軟硬體具體實施例之圖式; 圖3呈現出設定當成公司内部服務執行的本發明軟硬體 具體實施例之圖式; 圖4為依照本發明一個具體實施例,呈現出牵涉到電話 語音資料編碼與呈現的步驟之高階瀏覽流程圖; 圖5為依照本發明一個具體實施例,呈現由中介語音系 統執行用以接受與處理來電的過程之流程圖; 圖6為依照本發明一個具體實施例,呈現由中介語音系 統執行用以接受與處理來電的過程持續之流程圖; 圖7為依照本發明一個具體實施例的XML描述檔; 圖8為依照本發明一個具體實施例,描述用於接受語音 資料並且將它傳給適當編碼器進行壓縮的過程之流程圖; 圖9為依照本發明一個具體實施例,描述將語音資料編 碼成Qu i ckT i meTM Rea 1TM 或 W i ndows Med i aTM 格式的過程之 流程圖; 圖10為描述將&lt;311卜1^^6^編碼器初始化的次常式之流程 圖;479429 V. Description of the invention (6) Brief description of the drawings The present invention will be described with reference to the drawings. Its purpose is only for illustration and not limitation. The same reference numbers are used to indicate the same or corresponding parts. Among them: FIG. 1 is a diagram showing settings of many software and hardware components of the system according to a specific embodiment of the present invention; FIG. 2 shows specific embodiments of the software and hardware of the present invention performed by setting as an application service provider (ASP) Fig. 3 shows a specific embodiment of the software and hardware embodiment of the present invention that is set to be executed as an internal service of the company; Fig. 4 shows a step involved in encoding and presenting phone voice data according to a specific embodiment of the present invention High-level browsing flowchart; Figure 5 is a flowchart showing the process performed by the intermediary voice system to accept and process incoming calls according to a specific embodiment of the present invention; Figure 6 is a diagram showing the intermediary voice according to a specific embodiment of the present invention The flow chart of the continuous process of the system for accepting and processing incoming calls is shown in FIG. 7. FIG. 7 is an XML description file according to a specific embodiment of the present invention. FIG. 8 is According to a specific embodiment of the present invention, a flowchart of a process for receiving speech data and transmitting it to an appropriate encoder for compression is described. FIG. 9 is a description of encoding speech data into QuickT according to a specific embodiment of the present invention. Flow chart of the process of i meTM Rea 1TM or Windows Med i aTM format; FIG. 10 is a flowchart describing a subroutine for initializing a <311 1 ^^ 6 ^ encoder;

第9頁 479429 五、發明說明(7) 圖1 1為描述 圖; 圖1 2為描述 流程圖; 圖1 3為依照 碼成· w a v格式 圖1 4為描述 圖1 5為依照 過程的高階瀏 圖1 6為依照 資料出版的特 圖1 7為依照 語音資料檔位 圖1 8為依照 設備新增、編 流程圖。 將RealAudi〇TM編碼器初 ^Windows M e d i a™ 本發明 的過程 將現場 本發明 覽之流 本發明 定工作 本發明 置之文 本發明 輯和劉 一個具體實 之流程圖; 廣播排程的 一個具體實 程圖; 一個具體實 流程之流程 一個具體實 字取出與格 一個具體實 覽儲存在系 施例,施例,施例, 圖;施例, 式化過施例, 統資料 化 的次常式之流程 初始 化的次常式之 述將語音資料編 流程圖; $現通用工作流程 I現用於核准語音 #述將指出編碼過 程之流程圖; &amp;述使用網路前端 $内資料的過程之 較佳具體實施例之詳、纟-本發明是一種依照需求或以及時方^礎明 且透過電腦網路傳播之系統。本發明語Ϊ 回應伺服器(DIVR) 1〇8所構成,該伺服器配備有介面卡 4 1 0 6 ’可存取電話網路並且接收使用標準電話設備1 〇 4打出 的電話。D I V R 1 〇 8依照使用者描述播以及系統資料庫1 1 2 内含的其他分類資料作動,並且可執行工作流程並且出版 位於DI VR上的軟體,或者可為一部電腦。DIVR 108亦包含Page 9 479429 V. Description of the invention (7) Fig. 11 is a description diagram; Fig. 12 is a description flowchart; Fig. 13 is a coded · wav format; Fig. 14 is a description; Fig. 15 is a high-level browser according to the process; Figure 16 is a special chart published according to the data. Figure 17 is a file based on the voice data. Figure 18 is a flowchart of adding and editing according to the device. The implementation of the RealAudi〇TM encoder ^ Windows Medidia ™ The process of the present invention will present the present invention at a glance The present invention is scheduled to work The present invention is placed in the text invention series and a concrete flow chart; a specific implementation of broadcast scheduling Process diagram; a specific actual process flow a specific real word extraction and division a specific overview stored in the system examples, examples, examples, diagrams; examples, formalized examples, sub-normalized data The subroutine description of the initialization of the process will be the flow chart of the voice data; $ now the general workflow I now used to approve the voice # The description will indicate the flowchart of the encoding process; &amp; The comparison of the process of using the data in the network front end $ The details of the preferred embodiments are as follows-the present invention is a system that is based on demand or timely and spreads through a computer network. The invention is composed of a response server (DIVR) 108, which is equipped with an interface card 4 1 06 'that can access the telephone network and receive calls made using standard telephone equipment 104. D I V R 1 08 operates according to the user description and other classified data contained in the system database 1 12 and can execute the workflow and publish software on the DI VR, or it can be a computer. DIVR 108 also contains

第10頁 479429 五 、發明說明(8) 排 程程式以及載入軟體, 保存的語音資#。進入資】:播開始時廣播所 118 ^ !€i,JDTR〇^ (含)以上的編碼器12。…旦扭貝:將语音傳送給-個 資料流模體伺服器124即 ,之I,就會由 上的資料儲存梦署或者儲存在-個(含)以 上的貝t十係存裝置丨22内依照需求進行 * 此技藝的人士非常清楚,該組 對於精通 李统,戋者可π #田 μ、 3可位於單一電腦上的 糸統Α者了透過使用電腦網路(像 數個實體電腦進行散佈之系統。疋]際、周路)相連的複 請^閱圖1,此圖式呈現出本發 Γ乍者1G2使用連接到電話網路的電話 :活系^通常會依照脈衝代碼調變(pc 資料 該糸統透過介面裝置106連接至電話網路, 糸統可以接收來自電話網路的PCM資料。哼入^牡^舌 將電話按鍵音調訊號解碼,該訊號 &quot;衣置也可 (DTMF )訊妒。右一錄家办丨八品 )ϋ '疋已知的雙音調複頻 UMMH Λ唬有種乾例介面就是Dialogic^,造&amp;入&amp; 卡,其可連接到τ-1語音網路以及P0TS(老^ 路。該介面以插卡方式供應,安農 私。糸、,先)、,,罔 内。 文表於存取電話網路的電腦 此介面卡安裳於D0TELL互動式語音回應伺服器(divr) 108内,而DI VR 108内的軟體會接收並且確認欲存 的語音作者。DIVR 1〇8也藉由存取儲存在系統資料庫^2 上的使用者描述檔,確認浯音作者的識別碼。系 112為ODBC(開放式資料庫連結)相容資料庫,可由DIVR抖庫 479429 、發明說明(9) 1 08透過網路進行存取,並且由 是Sybase或心“丨#所出品的)#知式資料庫管理系統(像 五 是Sybaw我uracieu所出品的) DIVR 108會將工作流程以及王刺。 體1 1 0負責執行一系列步驟或規版,軟體11 0初始化’該軟 行的語音紀錄。每個使用者描述 M回應語音作者所進 行的特定步驟。在此執行的範』=義在錄製完成後要執 訊息,包含用電子郵件將錄f ^用於回應最新錄製的 員進行核准,或者透過FTP寄送遠:曰貧料連結寄送給管理 126,讓它包含於網頁中。而網的文字給網路伺服器 行的網路瀏覽器呈現給使用者/透過在用戶端PC1 28上執 而ίϊϊ::1 二t過網路連接到管理網路飼服器&quot;4, =中:ΪΓΛ設定成發出查詢並且更新到系、统 貝枓庫中以及將主現在網頁116上的 過此網路式前端11 4,李统管採g叮批v 卞^八化 逍 一 ^ ^乐、、·元3理貝可執行諸如新增使用者 和么司編輯現有使用者描述檔和組態以及預覽工作流程 輸出這些動作。系統資料庫112保存所有描述系統狀態以 及個別使用者喜好設定的資料,集中式使用者資料包含用 方;壓縮使用者语音資料的編碼器1 2 〇之名稱與位置、編碼 過·的浯音資料格式、是否授權使用者進行即時廣播以及使 用者在連結到寄給電腦網路上用戶端的錄製語音資料之前0 是否需要經過核准。 在DI VR 108確認作者並且開始錄製作業之後,di VR接受 語音資料並且將資料傳送給DTRouter 118 °DTRouter 118 接收來自D I V R 1 0 8分散區塊内的語音資料。依照特定語音Page 10 479429 V. Description of the invention (8) Schedule program and loading software, saved voice data #. [Entry information]: At the beginning of broadcasting, the broadcasting station 118 ^! € i, JDTR0 ^ (inclusive) or more encoder 12. … Once twisted: send the voice to a data stream motif server 124, that is, I, it will store the data from the above dream department or store it in a (inclusive) or more ten series storage device 丨 22 According to the needs of the people * This skill is very clear to those who are proficient in Lee ’s system, those who can be π # 田 μ, 3 those who can be located on a single computer through the use of computer networks (like several physical computers Dissemination system. 疋] Ji, Zhoulu) Connected to the reply please see Figure 1, this figure shows the original 1G2 phone using a telephone connected to the telephone network: live system ^ usually adjusted according to the pulse code The PC data is connected to the telephone network through the interface device 106, and the system can receive PCM data from the telephone network. Hum ^^^ tongue to decode the phone key tone signal, the signal can also be used (DTMF) jealousy. Right-hand recording home office 丨 eight products) 疋 '疋 The known two-tone multi-frequency UMMH Λ has a kind of dry interface is Dialic ^, made &amp; input &amp; card, which can be connected to τ -1 voice network and P0TS (Old ^ Road. This interface is supplied by card, Annong Private Shito first ,,) ,,, within indiscriminately. The text is displayed on the computer that accesses the telephone network. This interface is installed in D0TELL's interactive voice response server (divr) 108, and the software in DI VR 108 will receive and confirm the voice writer to be saved. DIVR 108 also accesses the user profile stored in the system database ^ 2 to confirm the identity of the sound author. Department 112 is an ODBC (Open Database Link) compatible database, which can be accessed through the network by DIVR dither library 479429, invention description (9) 1 08, and is made by Sybase or heart "丨 # 出品 的" # Knowledge-based database management system (such as the one produced by Sybaw I uracieu) DIVR 108 will work the workflow and the king of thorns. Body 1 1 0 is responsible for executing a series of steps or regulations, the software 11 0 initializes the voice of the soft line Records. Each user describes the specific steps that M responds to the author of the voice. The example performed here is to execute a message after the recording is complete, including e-mailing the recording f ^ to respond to the latest recording for approval. Or, send it via FTP: The poor link is sent to the management 126 so that it is included in the web page. The text of the web is presented to the user by the web browser of the web server / through the client PC1 28 上 执 and ϊϊ :: 1 Two t through the network to connect to the management network feeder &quot; 4, = Medium: ΪΓΛ is set to issue queries and update to the system, system database, and the main page 116 Of this network-type front end 11 4 Use gding batch v 卞 ^ 八 化 化 一 ^ ^ 乐 ,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,-,-,-,- The library 112 stores all the data describing the system status and the preferences of individual users. The centralized user data contains the users; the name and location of the encoder 1 2 〇, which encodes the user's voice data, the encoded audio data format, Whether to authorize the user to broadcast in real time and whether the user needs to be approved before connecting to the recorded voice data sent to the client on the computer network. After DI VR 108 confirms the author and starts the recording operation, di VR accepts the voice data and sends the data Send to DTRouter 118 ° DTRouter 118 receives voice data from DIVR 1 0 8 scattered blocks. According to specific voice

第12頁 479429 五、發明說明(10) 2的使用者描述槽,可指定一個(含)以上的編碼器120 音貢料。對於所指定的每部語音編碼器120而言, DTRc^ter }丨8會在缓衝區内建立一份所接收語音資料的副 本二DTRouter丨18 wTCP/ Ip(傳輸控制通訊協定/網際網路 通讯協$)socket連接開啟給每部編碼器12〇,並且在每次 接收到資料區塊或者緩衝區滿了時傳出資料。DTR〇uter 1 1 8根據使用者喜好設定資料做決定,並且將它傳遞當成 DIVR 1 〇8從系統貢料庫1 12内含資料產生包含χΜί資料的描 ,檔。下面將更清楚說明產生此XML描述檔的過程。系統 藉由將語音資料動態繞送到複數個以使用者直基 礎的編碼器,便可設定成最善加利用資源的;】 每部編碼器上執行負載平衡。事實上,也可將系統設定成 由包含使用者喜好設定資料的多重DTR〇uters 1 1 8所構成’或由負載平衡需求決定要使用哪部D 丁 R u t e r 118° 語音資料會傳遞給一個(含)以上的編碼器1 2 0進行壓 細。壓縮編碼器包含由R e a 1 N e t w 〇 r kTM I n c ·開發的 RealAudioTM壓縮/解壓縮程式(壓縮程式/解壓縮程式)、由 A p p 1 e C 〇 m p u t e rTM I n c ·開發的 Q u i c k T i m eTM 壓縮 / 解壓縮程 式、由Microsoft1^ Corp·開發的Windows MediaTM Format ‘ 或其他任何精通此技藝人士已知的壓縮編碼器。不過其缺 乏其他編碼器的壓縮優點,所以系統也可將資料存成WAV 格式。事實上,可將系統設定成運用任何數量或種類的編 碼器’如此同樣可讓系統資源得到最佳利用。Page 12 479429 V. Description of the invention (10) 2 User description slot, you can specify one or more encoder 120 tone materials. For each designated speech encoder 120, DTRc ^ ter} 丨 8 will create a copy of the received voice data in the buffer. DTRouter 18 wTCP / IP (Transmission Control Protocol / Internet) The communication protocol $) socket connection is opened to each encoder 120, and data is sent out every time a data block is received or the buffer is full. DTR〇uter 1 1 8 makes a decision based on user preference setting data, and passes it as DIVR 1 08 to generate a description and file containing χΜί data from the data contained in the system's database 1 12. The process of generating this XML description file will be explained more clearly below. The system can be set to make the best use of resources by routing the voice data to multiple user-based encoders;] Each encoder performs load balancing. In fact, the system can also be set up to consist of multiple DTR 0uters 1 1 8 containing user preference settings' or depending on the load balancing needs to determine which D Duter 118 ° voice data will be passed to one ( Including) The above encoders 1 2 0 are compacted. Compression encoder includes RealAudioTM compression / decompression program (compression program / decompression program) developed by Rea 1 N etw okrTM I nc, Q uick T developed by A pp 1 e C 〇mpute rTM I nc im eTM compression / decompression program, Windows MediaTM Format 'developed by Microsoft Corporation, or any other compression encoder known to those skilled in the art. However, it lacks the compression advantages of other encoders, so the system can also store data in WAV format. In fact, the system can be configured to use any number or kind of encoders' so that system resources can also be optimally used.

第13頁 479429 發明說明(11) 在將語音資料編碼之後,备 器124、資料儲存裝置m或這兩*丄至1貝料流媒體祠服 資料流媒俨朽服者’貢料儲存裝置122和 貝t十机媒體伺服器1 24兩者都位於網路 是否以及由哪個裝置將語音資 可置上。而 或資料流媒體㈣器124的^縮儲存裝置122 出版系統11〇就能存取這些資料 此工作流程以及 工作流程以及出版系統1丨〇會將任立 ^^^^λ^^„t,ν^νΛ θ #^^ 立次J τ 出版者會取得並將内含語 音育料位置的文字格式化。工作产葙糸试收取_ 1 3 作,爪权糸統將取得格式化的 連結亚透過FTP傳遞給網路飼服器126,網路飼服器126會 將連結動態結合到傳送給用戶端瀏覽器丨28的網頁中。^ 使用者透過用戶端瀏覽器1 2 8按下語音檔案連結存取網二 126時^網路伺服器會從位置上取得資料,或發出指令給 適當資料流媒體伺服器1 2 4將語音資料傳給用戶端。 圖2呈現本發明的另一個具體實施例。在此組態内,所 有系統組件由單一實體2 0 2操作。而在其他組態内,系統 組.件可分散至各處如此可達到最佳效能。此配置就是已知 的應用服務供應商或ASP。藉由控制系統的所有部份後, 控制公司可提供完整的語音資料流服務給任何數量的個人 或組織。 圖3呈現本發明的另一個具體實施例3 0 2。如此產生一種 情況,就是公司要在内部主控本發明的語音捕捉功能3 〇 2Page 13 479429 Description of the invention (11) After encoding the voice data, the backup device 124, the data storage device m, or both of these * 1 to 1 料 material stream media temple service data stream media corrupt servicer's tribute storage device 122 Both of the media servers 1 and 24 are located on the network whether and by which device the voice data is available. Or the data storage media device 124's ^ storage device 122 publishing system 11 can access these data. This workflow and workflow and publishing system 1 will ^^^^^ λ ^^ t, ν ^ νΛ θ # ^^ 立 次 J τ The publisher will obtain and format the text containing the location of the voice feed. Work industry trial collection _ 1 3 works, the claw right system will obtain the formatted link Asia Via FTP to the web server 126, the web server 126 will dynamically link the link to the web page sent to the client browser 丨 28. ^ The user clicks the voice file through the client browser 1 2 8 When the link access network 2 is 126, the network server will obtain the data from the location, or send an instruction to the appropriate data streaming server 1 2 4 to pass the voice data to the client. Figure 2 presents another specific implementation of the present invention For example, in this configuration, all system components are operated by a single entity 202. In other configurations, system components can be distributed everywhere so as to achieve the best performance. This configuration is a known application service Vendor or ASP. After controlling all parts of the system, The control company can provide a complete voice data stream service to any number of individuals or organizations. Figure 3 presents another specific embodiment of the present invention 302. This creates a situation where the company wants to control the voice capture of the present invention internally Function 3 〇 2

第14頁 479429 、發明說明G2) 此配置的一個優點是容易維護並且有能力依照特定公司的 需求自訂所有系統,像是獨特的隱密考量等等。此組態的 另—個優點是當系統核心功能在公司内部運作以及由公司 直接控制時,公司本身不需要將語音内容傳送給末端使用 者0 公司要在内部主控系統可透過D IV R、系統資料庫、編碼 器以及管理網路伺服器3 0 2加以控制。與儲存裝置和資料 流媒體伺服器相連的系統由系統許可裝置主控,或者由有 能力導引和壓縮語音來滿足公司需求之受信賴的第三廠商 裝置主控。這些裝置可位於系統可存取的任何網路上。 請參閱圖4,該圖式呈現出系統程式流程的高階概要。 語音作者使用電話設備4 0 2打電話給系統,而系統由 D IVR4 0 4應答。發話方的識別會對其在系統資料庫4 0 6内的 記錄進行認證,若作者提供的資料無效,D I VR將不理會通 話並且掛斷連接41 8,若作者的識別經過認證40 6,DIVR會 開啟一個連線到DTRouter並且傳送記錄的中繼資料40 8。 DIVR透過電話介面板41 0開始接收來自電話網路的語音資 料,並且將接收到的資料流傳給DTRou ter而到達41 2。 作者發出預定的DTMF音調414讓系統停止錄製,如此就 完成語音錄製。一旦錄製完成,D I V R會以和錄製及其排列❶ 4 1 6有關的資料更新資料庫。然後系統切斷通話4 1 8 ’同時 開始工作流程4 2 0,該流程會依照一組使用者定義的規則 與參數422執行。 回到步驟412,當DIVR接收資料後會將資料導引到Page 14 479429, Inventive Note G2) One advantage of this configuration is that it is easy to maintain and has the ability to customize all systems to the needs of a particular company, such as unique hidden considerations and so on. Another advantage of this configuration is that when the core functions of the system operate within the company and are directly controlled by the company, the company itself does not need to transmit the voice content to the end users. 0 The company's internal main control system can be accessed through D IV R, The system database, encoder and management web server 302 are controlled. The system connected to the storage device and the data streaming server is controlled by the system license device, or a trusted third-party device capable of guiding and compressing the voice to meet the company's needs. These devices can be on any network that the system can access. Please refer to Figure 4, which shows a high-level overview of the system program flow. The voice writer calls the system using the telephone device 402, and the system is answered by D IVR 404. The identification of the caller will authenticate its record in the system database 4 06. If the information provided by the author is invalid, DI VR will ignore the call and hang up the connection 41 8. If the author's identification is authenticated 40 6, DIVR It will open a connection to DTRouter and send recorded metadata 40 8. The DIVR starts to receive voice data from the telephone network through the telephone interface panel 41 0, and transmits the received data to DTRou ter to reach 41 2. The author issues a predetermined DTMF tone 414 to stop the system recording, thus completing the voice recording. Once the recording is complete, the D I V R will update the database with information related to the recording and its arrangement ❶ 4 1 6. Then the system disconnects the call 4 1 8 ′ and starts the work flow 4 2 0 at the same time. The flow will be executed according to a set of user-defined rules and parameters 422. Return to step 412, when the DIVR receives the data, it will direct the data to

第15頁 479429 五、發明說明(13) DTR0uter。DTRouter從DIVR或排程程式與載入程式系統接 收中繼以及語音資料424。在接收到資料之後,j)TR〇ut er 會開啟socket給由語音中繼資料定義之適當編碼器426。 資料經過開放式連接傳遞給編碼器428,該資料依照複數 2壓縮/解壓縮演算法執行語音資料的編碼43〇。編碼過的 語音會傳送給資料流媒體伺服器進行即時廣播432,或傳 給壓縮儲存裝置在稍後依照需求進行廣播434。然後媒體 伺服器會讓編碼過的資料變成可從網路存取43 6。該過 =最終結果是作者透過電話438所建立的語音資料^即^ 廣播或依照需求進行廣播。 明麥閱圖5 ,該圖式呈現由答盘處 作概要。一位级音作去广去金山、士 。蜒埋木包4的刼 的顧客前端&quot;7= 用傳統連接到電話網路 而# ®、“裳置(CPE ’例如電話)打電話502,m也藉由介 在if ΐί到電話網路的DIVRi在等待接電話504 ^接收到氣話時,會獲得發話號碼5〇6 娩碼51 0。若铐瑪1. 旨^&amp;個 的來電5〇4 糸統便會終止該通話並且等待新 靠著ί ^ ^放’將會提不作者輸入公司識別碼512。作者 收後合將= ϊ入識別資料|生贿,調514,在接 料敕i夺统=1曰9調所代表的號石馬。DIVR使用輪入的資 :否提供-個有效的公司 要、疋作者 公司碼,程式流程將回到牛㈣】9 1者輸入一個無效的 個機會提供有效的代碼。];=2在-在,作者會有另外- 右作者在二次内都無法輸入有效 479429 五、發明說明(14) 的公司識別碼,系統就會顯示n s 〇 r r y ’’訊息指出作者無法 提供有效公司碼5 2 0並終止通話,然後返回步驟5 〇 4等待接 收其他來電。 若作者提供了有效的公司識別碼,系統就會提示使用者 提供使用者識別碼522,這也是透過DTMF音調輸入524。而 所供應的識別碼一樣會與系統資料庫核對5 2 6,以決定作 者是否為有效的使用者5 3 0。跟之前一樣,使用者有三次 輸入有效使用者識別碼的機會。若無法輸入有效的識別Page 15 479429 V. Description of the invention (13) DTR0uter. DTRouter receives relays and voice data 424 from a DIVR or scheduler and loader system. After receiving the data, j) TRouter will open the socket to the appropriate encoder 426 defined by the voice relay data. The data is passed to the encoder 428 via an open connection. The data is encoded in accordance with the complex 2 compression / decompression algorithm 43. The encoded voice is sent to a streaming server for live broadcast 432, or to a compressed storage device for broadcast 434 at a later time as needed. The media server then makes the encoded data accessible from the network 43 6. The result = the final result is the voice data created by the author over the phone 438 ^ namely ^ broadcast or broadcast on demand. Mingmai reads Figure 5, which is presented as a summary by the answering desk. A graded music player went to Jinshan and Shishi. The customer's front end of the burial wooden bag 4 "7 = traditionally connected to the telephone network while the phone is called 502," Chang's (CPE ') such as a telephone), also through DIVRi to the telephone network via if Waiting to answer the call 504 ^ When you receive a gas call, you will get the calling number 5506 and delivery code 51 0. If the shame 1. The purpose of ^ &amp; an incoming call 504, the system will terminate the call and wait for a new call Writing ^ ^ ^ put 'will mention that the author enters the company identification code 512. After the author receives it, he will enter the identification information | Shima. DIVR uses round-robin funding: No provide-a valid company request, the company code of the author, the program flow will return to the cattle] 9 1 Enter an invalid opportunity to provide a valid code.]; = 2 In -in, the author will have another-the right author cannot enter a valid 479429 within the second time. 5. The company identification code of invention description (14), the system will display ns 〇rry '' message indicates that the author can not provide a valid company code 5 2 0 and terminate the call, then return to step 504 and wait for other incoming calls. If provided by the author A valid company identification code, the system will prompt the user to provide a user identification code 522, which is also input through DTMF tone 524. The supplied identification code will be checked with the system database 5 2 6 to determine whether the author is valid Of users 5 3 0. As before, the user has three opportunities to enter a valid user ID. If a valid identification cannot be entered

碼’系統就會顯示訊息指出作者無法提供有效使用者代碼 5 2 G並終止通話,然後返回步驟5 〇 4等待接收其他來電。 右提供有效的使用者代碼,系統會檢查資料庫決定使用 者是否超過所允許的最大通話數量532。若以經超過最大 通話數量,系統會顯示’’ sorry”訊息指出該事實52〇並且終 止通話,然後返回步驟5 0 4等待接收其他來電。若未超過 最大通話次數,系統會更新系統資料庫内作者的紀錄,以 ,應出已經打過新的通話次數5 3 4。系統也會決定使用者 是否要進行語音訊息現場分配53 6併發出適當的提示53 8和 540。然後將播放一般錄製指令542並且系統會等待〇了肝音 調544指出應該開始該錄製了。Code ’system will display a message stating that the author cannot provide a valid user code 5 2 G and terminate the call, then return to step 504 to wait for another incoming call. A valid user code is provided on the right. The system checks the database to determine if the user has exceeded the maximum allowed number of calls 532. If the maximum number of calls is exceeded, the system will display a "sorry" message indicating the fact 52 ° and terminate the call, and then return to step 5 0 4 to wait to receive other calls. If the maximum number of calls is not exceeded, the system will update the system database The author's record is that the number of new calls that have been made should be 5 3 4. The system will also determine whether the user wants to assign voice messages on-site 53 6 and issue appropriate prompts 53 8 and 540. Then the general recording instructions will be played 542 and the system will wait for a liver tone 544 indicating that it is time to start recording.

當作者提供0了乂?音調5 44指出錄製已經開始,1)1”合? :與:一心的連線546。D ϊ VR也會查詢系統資料庫二^ 立XML·描述檔,其中包含編碼、儲存以及導引電咭祖 料所需的所有參數548。一 η資料廑德门从i ° 、, 一貝科庫傳回作者的喜好設另 亚且建立XML描述檔,就會將它傳至DTR〇uter。 479429 五、發明說明(15) 該系統進行檢查以決定是否已經產生語音資料,即是決 定作者是否開始說話5 5 0。若作者正在說話,D I V R將接收 自電話網路5 52的PCM資料封包記錄起來574。系統也胯聽 D Τ Μ音調5 7 4指示訊息結束。原始P C Μ資料的副本儲存在本 機快取内5 5 4,所以作者可以複檢此訊息,而不用透過網 路從儲存裝置内取得。資料封包會送出給DTR〇uter556並 且控制會回到步驟5 5 0,在此系統在一次檢查下一個語音 資料封包。 當系統接收語音資料失敗時5 5 0,録製就會結束5 5 8。系 統決定作者是否暫停訊息56 0,若作者確實暫停訊息,系μ 統會等待562指示錄製繼續的DTMF音調5 64,或者強&quot;迫步驟 5 50上錄製程序繼續的逾時566。若錄製未暫停,系統將會 決定錄製是否完成5 6 8。若錄製未完成就結束,即是在步9 驟552内未接收到「結束錄製」DTMF音調,最有可能發1 掛斷或是線路中斷的問題57 2,然後系統會回到等$ ^尸 直到接收到其他來電。若接受到結束錄製音調::^ 系統會播放提示給作者以選擇錄製配置5 7 0。 、 圖6呈現決定已完成記錄的配置之過程。系統呈 =者選擇的配置選項6 0 2。#者藉由產生“ 的、擇之DTMF音調進行選擇604。若作者選擇回舜要 606,系統會在將PCM資料儲存在本機快取之二後,衣 料,然後重播配置選項6 08。若作者選擇忽:』播制放該段資 新錄製訊息61〇,則會將本機快取忽亚且重 自fi 1 9 Μ 平工且重新錄贺 心612,將控制交回步驟542。若作者決定儲存錄製6^说When did the author provide 0? Tone 5 44 indicates that the recording has begun, 1) 1 ”??: Connect with one heart 546. D ϊ VR will also query the system database 2 ^ Establish an XML · description file, which includes encoding, storage, and guidance electronics 咭All the parameters required by the ancestors are 548. An eta data will be transmitted from i °, i Bekoku back to the author's preferences, and an XML description file will be created, which will be transmitted to DTR〇uter. 479429 5 Description of the invention (15) The system checks to determine whether the voice data has been generated, that is, to determine whether the author started to speak 5 50. If the author is talking, DIVR records the PCM data packet received from the telephone network 5 52 and records it 574 . The system also listens to the D T M tone 5 7 4 to indicate the end of the message. A copy of the original PC M data is stored in the local cache 5 5 4 so the author can recheck this message without having to access the storage device via the network. Get. The data packet will be sent to DTR〇uter556 and control will return to step 5 50, where the system checks the next voice data packet at a time. When the system fails to receive the voice data 5 5 0, the recording will end 5 5 8. System decision Whether the author paused the message 5 0. If the author did pause the message, the system will wait for 562 DTMF tones 5 64 indicating the recording to continue, or force the recording program to continue overtime at step 5 50. 566. If the recording is not paused, The system will decide whether the recording is completed 5 6 8. If the recording is not completed, it means that the “End Recording” DTMF tone is not received in step 9 552. The most likely issue is 1 Hang up or the line is disconnected 57 2 And then the system will go back to waiting for $ ^ corpse until another call is received. If you receive the ending recording tone :: ^ The system will play a prompt to the author to select the recording configuration 5 7 0. Figure 6 presents the process of determining the configuration of the completed record. The system presents the configuration option 6 0 2 selected by the user. # 者 by selecting "DTMF tones" to choose 604. If the author chooses to return to Shun to 606, the system will store the PCM data in the local cache two, and then replay the configuration option 6 08. If The author chooses to ignore: "Play and release the new recording message 61. This will cache the local player and re-start it from fi 1 9 Μ Ping Gong and re-record the heart 612, and return control to step 542. If The author decides to save the recording

479429 、發明說明(16) 則貪料庫内指示錄製數量的資料將會遞增6 1 6,而錄製的 名稱和位置會儲存在資料庫内61 8,並且作者的工作流程 將執行6 2 G。系統將掛斷連線6 2 2並且將控制交回步驟 5 0 ^。最後’將指示作者要忽視錄製並且結束退出6 2 4,如 此系統會忽略本機快取並且掛斷連線626,將控制交回步 驟 5 0 4 〇479429, invention description (16), the number of recorded materials in the greed database will be increased by 6 1 6 and the name and location of the recording will be stored in the database 61 8 and the author's workflow will execute 6 2 G. The system will hang up the connection 6 2 2 and return control to step 5 0 ^. Finally ’will instruct the author to ignore the recording and exit 6 2 4. If this system ignores the local cache and hangs up the connection 626, return control to step 5 0 4 〇

當DI VR查詢系統資料庫,並且在圖5步驟548内取得一組 預定語音作者參數時,將會產生XML描述檔。該資料負責 驅動所有有關作者語音資料錄至與儲存之系統決定。圖7 王現一個範例XM L描述檔,如同本發明一個具體實施例所 預期的。所有資料都包含在一組&lt;ENC〇DELIST&gt;標籤内 702,而就在&lt;ENCODELISTM$籤開頭上是用於識別語音資 料704作者的公司與使用者識別碼之標籤。其中也有設定 等待與優先權值的標籤,此後將進行討論,其在排定現場 廣播時使用。〈SOURCE〉區塊70 6說明系統接收的語音資料 格式。來源格式或原始語音資料像什麼會以其代崎、取樣 率以及大小(多少位元組)來做說明。大多數電話系統使用 壓細過的// L a w和a L a w格式廣播資料,系統會確認接收該 資料並且將它解壓縮,產生線性PCM資料。 XML描述檔的剩餘部份由複數個〈ENCODE〉區塊70 8構成。 圖7内呈現的範例X M L描述樓分別由三個編碼區塊構成,— 個是Sun Audio、一個是RealAudio 另一個是WindowsWhen the DI VR queries the system database and obtains a set of predetermined speech author parameters in step 548 of FIG. 5, an XML description file will be generated. This data is responsible for driving all system decisions regarding the recording and storage of the author's voice data. Figure 7 shows an example XM L profile, as expected by a specific embodiment of the present invention. All the data is contained in a set of &lt; ENCODELIST &gt; tags 702, and just before the &lt; ENCODELISTM $ signature is a tag for identifying the company and user identification code of the author of the voice data 704. There are also tags that set wait and priority values, which will be discussed later, and are used when scheduling live broadcasts. <SOURCE> Block 70 6 describes the format of the voice data received by the system. What the source format or raw audio data looks like will be described in terms of its generation, sample rate, and size (how many bytes). Most telephone systems use the compacted // L aw and a L aw formats for broadcast data. The system will confirm receipt of the data and decompress it to produce linear PCM data. The remainder of the XML description file is composed of a plurality of <ENCODE> blocks 708. The example XML description building presented in Figure 7 consists of three coding blocks, one is Sun Audio, one is RealAudio, and the other is Windows.

Med i a。每個區塊以編碼器欲產生的媒體名稱做開始,緊 接著是用於執行即時廣播的檔案名稱。在此也有一個用 '於Med i a. Each block starts with the media name that the encoder wants to generate, followed by the file name used to perform live broadcast. There is also a

479429 五、發明說明(17) 壓縮編碼,並包含壓縮/解壓縮程式名稱的〈c 〇 D E c L〖$ τ〉區 塊710。在Real Audio的情況中,單獨資料流可使用多個壓 縮/解壓縮程式以多重比率進行編碼。然後資料流伺服器 和用戶端將協議用可以成功傳送過使用者連線的最快速率 (例如最咼品質)。在此也提供用於執行即時廣播的資料节 伺服器位址,以及用於儲存語音資料壓縮副本的儲存裝 置。使兩XML所提供的彈性允許特定作者產生的語音資料 繞送到特定伺服器’有可能是一個最靠近的伺服器、位於 較不擁塞的網路區段上或者根據作者的識別碼而有較的479429 V. Description of the invention (17) Compression coding and <c o D E c L [$ τ>] block 710 containing the name of the compression / decompression program. In the case of Real Audio, a single data stream can be encoded at multiple ratios using multiple compression / decompression programs. The data stream server and client then use the protocol at the fastest rate (such as the highest quality) that can successfully pass the user connection. Data section server addresses for performing live broadcasts, and storage devices for storing compressed copies of voice data are also provided here. The flexibility provided by the two XMLs allows voice data generated by a particular author to be routed to a particular server. 'It may be the closest server, located on a less congested network segment, or more based on the author's identification code. of

請參閱圖8,此圖式呈現經常接收語音資料給編碼器的 DTRouter過程。一旦伺服器初始化8〇2,其會處於等待狀 態8 0 4直到接收到連線。當接收到來自d I v r的連線8 〇 6, XML描述檔會說明首先接收到的語音資料以及編碼參數。 DTRouter解析XML描述檔8 07,並且決定是否附加於準則規 格内80 8。若XML描述檔包含異常資料,則DTR〇uter會刪除 該作業並且返回步驟804並等待其他連線。 示Please refer to FIG. 8, which illustrates a DTRouter process that frequently receives voice data to the encoder. Once the server has initialized 802, it will wait for 804 until it receives the connection. When a connection 806 from d I v r is received, the XML description file will describe the speech data and encoding parameters received first. DTRouter parses the XML description file 8 07 and decides whether to attach it to the standard specification 80 8. If the XML description file contains abnormal data, DTRuter will delete the job and return to step 804 and wait for other connections. Show

若XML描述檔有效,則資料會載AXMUnf〇物件81 〇,該 物件為將資料傳遞給編碼器的資料結構。在此將創造 / XMLInfo物件給XML描述檔内指定的每個編碼器,並且執行 才双查以決疋接收到的X M L描述檔是否與存在的工作互相於 配’該工作目前由DTR0U ter處理複數個語音通話8丨2内&amp; 行的任何緒所處理。若X M L描述檔與現有工作未搭配,則 控制會交給最新創造的消費者緒8 1 4,其負責接收進入的If the XML description file is valid, the data will contain the AXMUnf0 object 81 0, which is a data structure that passes data to the encoder. Here, an XMLInfo object will be created for each encoder specified in the XML description file, and a double check will be performed to determine whether the received XML description file matches the existing job. The job is currently handled by DTR0Uter. Any voice call within 8 & 2 will be processed. If the X M L profile does not match the existing job, control will be passed to the newly created consumer thread 8 1 4 who is responsible for receiving incoming

479429 五、發明說明(18) 語音資料。DTRouter也建立緒以將接收的資料傳送給作者 XML描述檔8 1 6内指定的編碼器。在已經建立適當緒並且將 控制交給它們之後,主緒會回到步驟8 〇 4並且等待下一個 連線。 消費者緒試圖讀取來自開放STCP/IP s〇cket以XML描述 檔8 1 8做開始的資料,在此將會進行檢查看看是否可取得 資料82 0,例如D IVR會傳送捕捉自語音作者的pcm資料。若 資料存在,則會從socke t讀取資料8 22。該資料會讀取並 置於緩衝區内8 2 4,並且將訊號傳送至可獲得資料的編碼 器緒826。之後控制會交回步驟820看看是否還可獲得其他 資料區塊。如果已經沒有資料了 ,錄製將會終止並且刪除 消費者緒838。 一群編碼器緒會平行操作,一個緒用於每個XML描述檔 指定的編碼器以便將語音資料編碼。編碼器緒開&amp;TCp socket給編碼器82 8,一旦連線已經開啟,將會檢查消費 者緒是否已經產生資料可用的訊號8 3 0。若可獲得資料, 將會從緩衝區讀取832並且透過socket寫給編碼器834。一 旦已經寫入資料,編碼器緒會發出訊號給可獲得資料緩衝 區的消費者緒8 3 6。之後控制會交回步驟83 0,在此會進行 檢查看是否有下一個資料區塊。如果已經沒有資料了,錄 製將會終止並且刪除編碼器緒8 3 8。 每個執行中的消費者緒都由其處理的XML描述檔識別, 可將其看成交回步驟812的峰一指紋,現有的XML描述檔可 與DTRouter内複數個消費者緒所處理的XML描述檔相比479429 V. Description of the invention (18) Voice materials. DTRouter is also set up to send the received data to the encoder specified in the author's XML description file 8 16. After the appropriate threads have been established and control is passed to them, the main thread will return to step 804 and wait for the next connection. The consumer attempted to read the data from the open STCP / IP socket starting with the XML description file 8 1 8 and will check to see if the data 8 2 0 is available. For example, D IVR will send a capture from the voice author Pcm information. If the data exists, the data will be read from socks 8 22. The data is read and placed in the buffer 8 2 4 and the signal is sent to the encoder 826 where the data is available. Control then returns to step 820 to see if other data blocks are available. If there is no more data, the recording will be terminated and consumer thread 838 will be deleted. A group of encoder threads operates in parallel. One thread is used for the encoder specified in each XML description file to encode the speech data. The encoder opens &amp; TCp socket to the encoder 82 8. Once the connection has been opened, it will check whether the consumer has generated a signal 8 30 that the data is available. If information is available, 832 will be read from the buffer and written to the encoder 834 via the socket. Once the data has been written, the encoder thread will send a signal to the consumer who has access to the data buffer area 8 3 6. Control will then return to step 8300, where it will be checked to see if there is the next data block. If there is no more data, the recording will be terminated and the encoder thread will be deleted. Each executing consumer thread is identified by its processed XML description file, which can be viewed back to the peak-fingerprint of step 812. The existing XML description file can be compared with the XML description processed by multiple consumer threads in DTRouter. Compare

第21頁 4ff9429 五、發明說明(19) 擬。若系統定義同等XML描述檔(例如包含一致的公司與使 用者識別碼)由目前執行的緒處理,在8 4 0内會進行檢查, 看看最新接收的XML描述檔是否有由其〈PRIORITY〉標籤7〇4 識別的較高值。若接收到的XML描述檔沒有較高的優先 權’則DTRouter會忽視該資料並且返回步驟8〇4等待新的 連線。若XML描述檔有較高的優先權,預先存在的消費者 緒就會暫停8 4 2並且建立新的消費者緒8 4 4。 因為現有的消費者緒會與編碼器緒一起合作,所以最新 建立的消費者緒844就會持續傳遞資料給它。在846上進行 檢查’以決定最新接收的XML描述檔是否指定與預存緒處 理的XML描述檔(之前在步驟842内已經暫停)不同的壓縮位 置。若已經指定新壓縮位置,則會建立額外的編碼器緒以 接收來自消費者緒的資料,以便編碼到指定壓縮檔内 8 4 8。若未指定新的壓縮檔8 4 6,則只有現有的編碼器可對 進入的語音進行編碼並持續使用新建立的消費者緒。稍後 將做解釋,此常式會用於即時廣播語音的排程以及載入。 圖9主現由设疋成將資料編碼為如丨以丁丨託、Reai或 Windows Media格式的伺服器執行之處理。當伺服器一開 始處理時,會將跨聽socket初始化902,其用於等待來自 DTRouter的連線90 4。當連線建立後,將會利用連線特定 緒(由主緒9 0 8生產的)從s 〇 c k e t 9 0 6讀取資料,然後主 會回到步驟9 0 4等待下一個連線。 、 連線特定緒會接收XML描述檔,該檔附加到語音資料去 成“?€ 9 1 〇。X M L描述檔經過解析9 1 2,並且編碼器決定此Page 21 4ff9429 V. Description of Invention (19). If the system defines the equivalent XML description file (for example, it contains consistent company and user identification codes), it will be processed in the current thread, and it will be checked in 840 to see if the newly received XML description file has its <PRIORITY> Higher value identified by label 704. If the received XML description file has no higher priority, DTRouter will ignore the data and return to step 804 to wait for a new connection. If the XML description file has higher priority, the pre-existing consumer thread will be suspended 8 4 2 and a new consumer thread 8 4 4 will be created. Because existing consumer threads work with encoder threads, the newly created consumer thread 844 will continue to pass data to it. A check is made on 846 'to determine if the newly received XML description file specifies a different compression location from the pre-stored XML description file (which was previously paused in step 842). If a new compression location has been specified, an additional encoder thread will be created to receive data from the consumer thread for encoding into the specified compression file 8 4 8. If no new compression file 8 4 6 is specified, only the existing encoder can encode the incoming speech and continue to use the newly created consumer thread. As explained later, this routine is used for scheduling and loading of real-time broadcast voice. FIG. 9 is a process performed by a server configured to encode data into a format such as DingTuo, Reai, or Windows Media. When the server starts processing, it will initialize 902 the listening socket, which is used to wait for the connection 90 4 from the DTRouter. After the connection is established, the connection specific thread (produced by the main thread 908) will be used to read data from sock e t 9 0 6 and then the host will return to step 9 0 4 and wait for the next connection. The connection specific thread will receive an XML description file, which is attached to the voice data to become "? € 9 1 0. The X M L description file is parsed 9 1 2 and the encoder decides this

第22頁 479429Page 479 429

處包含的資料是否包含有效參數可對進來的語音資料進行 編碼914。在描述檔内含的XML資料無效之處,會將緒刪除 亚且關閉由其維護的連接91 6。若資料有效,編碼器會初、 始化918並且從socket讀取語音資料92〇。一旦開始從曰 DTRout er開始傳輸語音資料,則會從s〇cket讀取該資料並 且由壓縮/解壓縮程式進行編碼9 2 4。若壓縮語音資料的編 碼is是Qu 1 ckT 1 me編碼器,一旦所有編碼完成後則合在編 石馬的電影樓中附加提示軌92 6。在編碼完成後編碼^將會 關閉9 2 8,編碼過的語音資料會依照XML描述檔9 3 〇儲存^Whether the data contained here contains valid parameters can encode the incoming voice data 914. Where the XML data contained in the profile is invalid, the thread will be deleted and the connection maintained by it will be closed. If the data is valid, the encoder will initialize, initialize 918 and read the voice data 92 from the socket. Once the transmission of voice data from DTRouter begins, the data will be read from the socket and encoded by the compression / decompression program 9 2 4. If the code of the compressed speech data is a Qu 1 ckT 1 me encoder, once all encoding is completed, it will be combined in the movie building of Shima to add a cue track 92 6. After encoding is completed, ^ will be closed 9 2 8 and the encoded voice data will be stored according to the XML description file 9 3 〇 ^

資料儲存裝置或資料流媒體伺服器内。連線特定緒停止並 且其與DTRouter的連線將會關閉91 6。 圖1 0呈現使用Qu i ckT i me編碼器壓縮進入的語音資料之 程序。在1 00 2内先設定壓縮/解壓縮程式,然後語音磁執 就可準備將進來的資料寫入1 004,之後代碼會設定成準號 接收資料進行編碼的狀態1 0 0 6。這樣就完成初始化並且^ 束次常式1008,將控制交回主常式。Data storage device or streaming server. The connection is stopped and its connection to DTRouter will be closed 91 6. Figure 10 presents the procedure for compressing the incoming speech data using a Quick coder. Set the compression / decompression program in 00 2 first, and then the voice magnetic preparation can prepare to write the incoming data to 1 004, and then the code will be set to the standard number. The status of receiving data for encoding is 1 0 0 6. This completes the initialization and ^ bundles subroutine 1008, returning control to the main routine.

圖11呈現使用Real Audio編碼器壓縮進入的語音資料之 程序。編碼器首先決定XML描述檔是否指示編碼器已超過 一個壓/解壓縮程式的方式對語音資料進行編碼11〇2,若 扎不使用單一壓縮/解壓縮程式,則會進行初始化丨丨〇 6。 在指示使用多個壓縮/解壓縮程式之處,將會為每個壓縮/ 解壓縮程式進行SureStream初始化丨丨04。SureStream是Figure 11 shows the procedure for compressing incoming voice data using a Real Audio encoder. The encoder first determines whether the XML description file indicates that the encoder has more than one compression / decompression program to encode speech data. If it does not use a single compression / decompression program, initialization will be performed. Where multiple compression / decompression programs are instructed, SureStream initialization will be performed for each compression / decompression program. SureStream is

Real Networks所研發的一種專利技術,其可用多種壓縮/ 解壓縮程式對進入的語音資料進行編碼並且儲存在單一檔A patented technology developed by Real Networks that encodes incoming voice data with multiple compression / decompression programs and stores it in a single file

/^29 五、發明說明(21) &quot;一~ 一 ' ' 木内。然後用戶端要求連線,S u r e S t r e a m開始協調伺服哭 與用戶端,使用用戶端連結所支援的最佳編碼傳送資料 /¾. 〇 。、此過程持續決定語音是否即時廣播i丨0 8,這同樣由編碼 器接收的XML描述檔決定。若指示要即時廣播,伺服器會 將資料流媒體伺服裔軟體初始化,以直接接收和廣播其編 碼的語音資料1 π 〇,然後編碼器會決定語音是否要壓縮到 儲存裝置内1 1 1 2,若要壓縮語音,則會將輸出檔案位置初 始化1 Π4,然後編碼引擎會設定成已經準備開始編碼 U巧,並且完成初始化丨丨丨8。若指示不進行壓縮,則會 壓縮/解壓縮程式設定成準號開始編碼1116,並且 9 、 始化11 18。 凡战初 、,圖η呈現使用Windows Media編碼器壓縮進入的語音資 ,之程序。在步驟丨2〇2内將壓縮/解壓縮程式初始化,二 L :碼器會決定語音是否要即時廣播12〇4。若指示要即、時 二U服器會執行在伺服器所用檔案目錄徑有附加別名 的育料流媒體伺服器軟體12〇6,然後編碼器會決定往if I;厂=健:裝置内12。8,若要壓縮語音,則會; 始二= 成已經準傷開 x且凡成初始化1214。右指示不進行壓縮, 則會將壓縮/解壓縮程式設定成準號開始編碼1 2 1 2,並且 完成初始化1 2 U。 請參閱圖13 ’此圖^呈現將標準未M縮的.w 編碼之過程。t —開始時㈣器將語音資料儲存成/ ^ 29 V. Invention Description (21) &quot; 一 ~ 一 '' Kinai. Then the client asks to connect, and Su r e s t r e a m starts to coordinate the servo cry with the client, using the best encoding supported by the client connection to send the data / ¾. 〇. 2. This process continuously determines whether the voice is broadcasted in real time, which is also determined by the XML description file received by the encoder. If instructed to broadcast in real time, the server will initialize the streaming server software to directly receive and broadcast its encoded voice data 1 π 〇, and then the encoder will determine whether the voice should be compressed into the storage device 1 1 1 2 If you want to compress the speech, the output file position will be initialized to 1 Π4, and then the encoding engine will be set to be ready to start encoding, and the initialization will be completed. If it is instructed not to compress, the compression / decompression program is set to a semicolon number to start encoding 1116, and 9, 1111 is initialized. At the beginning of the war, Figure η shows the procedure for compressing the incoming voice data using Windows Media encoder. The compression / decompression program is initialized in step 丨 202, and the L: coder determines whether the voice is to be broadcasted in real time. If instructed to do so immediately, the U server will run the breeding stream media server software 1206 with an additional alias in the file directory path used by the server, and then the encoder will decide to go to if I; factory = Jian: 12 .8, if you want to compress the voice, it will; The first two = Cheng has already broken x and Fancheng initialized 1214. The right indicates that no compression is performed. The compression / decompression program will be set to the reference number to start encoding 1 2 1 2 and the initialization 1 2 U will be completed. Please refer to FIG. 13 ′ This figure ^ presents the process of encoding a standard uncompressed .w. t — At the beginning the voice data is stored as

479429 五、發明說明(22) 案’socket會初始化1302並等待接受來自訂R〇uter的連線 1 304。當連線建立後,將會利用連線特定緒(由主緒丨3 〇6 生產的)接受橫跨s 〇 c k e t 1 3 0 8的資料。然後主緒會回到步 驟1304等待下一個連線。 連線特定緒會接收XML描述檔,該檔附加到語音資料當 成標題1310 cXML描述檔經過解析1312,並且編碼器決定 此處包含的資料是否包含有效參數可對進來的語音資料進 =瑪1314。在XML描述檔無效之4,將會刪除緒並且關 閉由^緒造成的連線1316。若資料有交文,編碼器會準備 v彳:題仁是不供應數值給檔案内含的資料長度1 3 8。 、扁碼器從開放式socket讀取資料13?0 a、说 41 322,則會將該資料寫入· wav資料 一 B p ^拉 收所有資/料,編碼器會將資料長产:挪 ·至接 檔宰132fi。名紝$ +丄貝丁十负度冩入· wav標題並且關閉 子田木U26。在結束完成後,編碼過立次479429 V. Description of Invention (22) Case ’socket initializes 1302 and waits to accept connection 1 304 from Router. When the connection is established, it will use the connection specific thread (produced by the main thread 3 06) to accept data across s 0 c k e t 1 3 0 8. The thread then returns to step 1304 and waits for the next connection. The connection specific thread will receive an XML description file, which is attached to the voice data as the title 1310. The cXML description file is parsed 1312, and the encoder determines whether the data contained here contains valid parameters to input the incoming voice data = Ma 1314. Invalid XML profile 4, the thread will be deleted and the connection 1316 caused by the thread will be closed. If there is a text in the data, the encoder will prepare v 题: The title is not supplied with the value of the data length 1 3 8 contained in the file. The flat coder reads the data from the open socket 13.0 a, say 41 322, it will write the data · wav data-B p ^ pull all the data / materials, the encoder will produce data long: · To the stall to slaughter 132fi. The name 纴 $ + 丄 bedin ten negative degrees entered the wav title and closed Kodagi U26. After finishing the coding

描述擋儲存在資料儲存梦署$次u …曰貝科會依照XML 連線特定緒停止Γίί二置媒體飼服器内1 32 8。 本發明提供—種$ = 0Uter的連線將會關閉1 31 6。 能力。圖“内、所:過電腦網路即時語音電話廣播的 行的處理負責廣播在幾=入程式(S/L系統)執 廣播。系'統資料庫储存有:程;立即進 疋保存語音資料已經排程以及结束^事件的_貝料,尤其 經在2 : 00ΡΜ排程開始現場廣播,系例如,若使用者已 :直到作者開始自己廣播為止。不、過:f始廣播保存的語 者無法進行排程廣#,若作者在細J ;有一種情況就是作 479429The description file is stored in the data storage dream department $ 次 u… said that Beco will stop in the two media feeders according to the XML connection specification 1 328. The present invention provides that a connection of $ = 0 Uter will be closed 1 31 6. ability. Figure "Inside and outside: the processing of the real-time voice call broadcast via the computer network is responsible for broadcasting in the program (S / L system) to perform the broadcast. The system's database stores: procedures; immediately save voice data The _ shell material that has been scheduled and ended ^ event, especially started live broadcasting at 2:00 PM schedule, for example, if the user has: until the author starts broadcasting on his own. Cannot perform schedule Guang #, if the author is in fine J; there is a case of 479429

始線傳輸 輸0 事件伴隨的資料會指示系統終止保存語音的傳 j程序會持續迴圈進行並輪詢系統,決定需要 扣曰廣播直到現場作者準號開始廣播之排程的現場廣播= 否開始。迴圈起始於S/L系統查詢系統資料庫,決定〃已: 排程與下-分鐘内開始的事件14〇2。若在目前使用者開: 一個已經排程的現場廣播14〇4,S/L系統將產生一個使用° 者XML描述檔複本1 40 6並且將它傳送給DTRouter 14〇8。 系統資料庫儲存排程事件隨附的語音資料,例如一個南 用的「等待開始傳輸」訊息或給特定使用者的訊息。在$ 送XML描述檔之後1 408,S/L系統會存取本機或網路檔案系 統1 41 2以開啟該排程事件隨附的語音資料丨4丨〇。一旦開” 啟,該語音資料就會傳送到DTRouter 1414。然後系統會 查詢資料庫,決定在目前是否有其他事件排定要開始-141 6。若有其他事件排定要開始141 8,則會從步驟14〇6重 新執行廣播音樂的過程。 若沒有事件排定要開始1 4 1 8和1 4 0 4,系統會再一次杳古旬 資料庫取得已經排定要在目前結束的事件資料1 422。在目 前.並沒有事件要結束之處1 424,系統會等待一分鐘142〇並 且開始結束處理1 4 0 2。若資料庫傳回指示保存語音在結束 的消息1 424,系統會產生一個負優先權等級的事件XML描 述棺,用於指示欲終止廣播,然後X M L描述檔會傳送到 D T R 〇 u t e r 1 4 2 8。在此產生一個查詢取得已經排定要在此 時結束的額外事件之資料1 4 30若資料庫傳回指示保存語音The data accompanying the 0 event will instruct the system to terminate the transmission of the stored voice. The j program will continue to loop and poll the system. It is determined that the broadcast needs to be deducted until the live author's license number starts the scheduled live broadcast = no start . The loop starts when the S / L system queries the system database and decides whether it has been: Scheduled and events starting in the next minute 1402. If the current user opens: A scheduled live broadcast 1404, the S / L system will generate a copy of the XML description file 1 40 6 and send it to DTRouter 14 0. The system database stores voice data attached to scheduled events, such as a Southern "waiting to start transmission" message or a message to a specific user. After sending the XML description file at 1 408, the S / L system will access the local or network file system 1 412 to open the voice data attached to the scheduled event. Once turned on, the voice data will be transmitted to DTRouter 1414. The system will then query the database to determine if there are currently other events scheduled to start -141 6. If there are other events scheduled to start 141 8 then Re-execute the process of broadcasting music from step 1406. If there is no event scheduled to start 1 4 1 8 and 1 4 0 4, the system will once again obtain the event data 1 that has been scheduled to end at the current time. 422. At the moment, there is no event to end at 1 424, the system will wait for a minute 1420 and begin to end processing 1 2 0 2. If the database returns a message 1 424 indicating that the saved voice is at the end, the system will generate a The event description of the negative priority XML description coffin is used to indicate that the broadcast is to be terminated, and then the XML description file will be transmitted to DTR 〇uter 1 4 2 8. A query is generated here to obtain additional events that have been scheduled to end at this time. Data 1 4 30 If the database returns instructions to save the voice

第26頁 479429 五、發明說明(24) 要在此時結束的資訊1 4 3 2,則會從步驟1 4 2 6 保存語音廣播的處理。在目前並沒有事件要 1 432,系統會等待一分鐘1 4 20並且開始結束 如同之前所解釋的,DIVR會初始執行一系 用者的預定動作或事件之工作流程軟體。該 一系列以像是PERL或PYTHON這些指令語言寫 一系列以C/C + +語言寫成的組譯可執行程式, 的組合。在此執行該工作流程以回應語音作 (請參閱圖6步驟6 2 0 )。 圖1 5依照本發明的一個具體實施例,呈現 作流程。執行工作流程的系統會將一系列參 流程軟體1 5 0 4,工作流程軟體接收到參數之 些參數在系統資料庫内執行查詢1 5 0 6,並且 隨附的事件1502。而在1508内也會設定工作 態,避免同時執行多個相同工作流程複本。 體也會取得其他中繼資料1 5 1 0以及使用者資 資料與正確執行工作流程所需的紀錄有關。 含工作流程1 5 1 4和1 5 1 6的事件,直到所有事 過為止。在所有事件完成之後,工作流程便 圖1 6依照本發明的一個具體實施例,呈現 程的特定具體實施例。當一位作者透過系統 (他必須要經過管理員的核准),就會執行該 程。在錄製完成之後就會執行工作流程1 6 0 2 封電子郵件給管理員,說明網頁的超連結内 重新執行終止 結束之處 處理1 4 0 2。 列適合個別使 工作流程包含 成的指令碼、 或者這兩者 者完成的通話 出一個通用工 數傳遞給工作 後,會使用這 取得工作流程 流程的執行狀 該工作流程軟 料1 5 1 2,這些 該軟體執行包 件都已經執行 會終止1 5 2 0。 出一個工作流 紀錄通話時 範例工作流 ,然後寄發一 已經有一個紀Page 26 479429 V. Description of the invention (24) The information to be ended at this time 1 4 3 2 will save the processing of the voice broadcast from step 1 4 2 6. At present there are no events that require 1 432. The system will wait for 1 4 20 and start and end. As explained earlier, DIVR will initially execute a workflow software for a series of user-defined actions or events. This series is a combination of a series of executable executable programs written in C / C ++ language, such as PERL or PYTHON. The workflow is executed here in response to the voice action (see step 6 2 0 in FIG. 6). Figure 15 presents a workflow according to a specific embodiment of the invention. The system that executes the workflow will run a series of parameters of the software 1504, and the parameters received by the workflow software will execute the query 1506 in the system database, and the incident 1502 will be attached. In 1508, the working state will also be set to avoid executing multiple copies of the same workflow at the same time. The organization will also obtain other metadata 1 5 10 and user information related to the records required to properly perform the workflow. Events with workflows 1 5 1 4 and 1 5 1 6 until everything is over. After all events are completed, the workflow is as shown in Figure 16. According to a specific embodiment of the present invention, a specific embodiment of the process is presented. When an author goes through the system (he must be approved by the administrator), the process is performed. After the recording is completed, the workflow will be executed 1 6 0 2 e-mails to the administrator, explaining the re-execution termination in the hyperlink of the web page, the end of the processing 1 2 0 2. The column is suitable for the script code that makes the workflow included, or a call that is completed by both of them. After passing a common number to the job, it will use this to obtain the execution status of the workflow process. The workflow material 1 5 1 2 These software execution packages have been executed and will terminate 1 520. Create a workflow, record a sample workflow during a call, and send a

第27頁 479429 五、發明說明(25) 錄1 6 0 4。該工作流程藉由從系統資料取的資料來產生訊 息。 管理員點選電子郵件訊息内的超連結1 6 0 4導覽到管理網 頁1 6 0 5,管理網頁允許管理員設定已紀錄語音資料的屬 性,像是標題命名以及設定其「核准」狀態。管理員會將 紀錄的標題命名1 6 0 6,並且檢查其有效性1 6 0 8。若標題無 效,管理員會回到步驟1 6 0 5内呈現的管理頁面,並且提供 機會再次為紀錄命名。若標題有效,系統資料庫内錄製的 紀錄將會更新1 6 1 0,然後核准者會設定錄製的「核准」屬 性1 6 1 4。若錄製並未核准1 6 1 6,則核准者會呈現彙總對於 資料庫内錄製紀錄所做變更之畫面1 6 1 8,並且終止工作流 程 1 62 0。 當管理員核准錄製時1 6 1 6,將會更新資料庫以反映核准 狀態的改變1 6 2 2,然後會同時執行三個動作。將錄製的副 本當成電子郵件附件傳送給特定的檔案保管人1 6 2 6,管理 員核准錄製後會出現一個彙總對資料庫内錄製紀錄所做的 變更之晝面1 6 1 8,然後也會執行稍後說明的出版者1 6 2 4。 出版者的成果會透過FTP傳遞到特定網路伺服器,而包含 於網頁成為其一部份1 6 2 8。在這三個步驟執行完畢之後, 工作流程就會終止1 6 2 0。 出版者也是一套由DIVR初始化的軟體子系統,用於產生 指示語音資料位置是在網路儲存裝置上、執行現場廣播的 資料流伺服器或這兩者上的文字資料。就像工作流程軟體 一樣,出版者包含一系列以像是PERL或PYTHON這些指令語Page 27 479429 V. Description of the Invention (25) Record 1 6 0 4 The workflow generates information from data taken from system data. The administrator clicks the hyperlink 1 600 in the email message to navigate to the management web page 16 0 5. The management webpage allows the administrator to set the properties of the recorded voice data, such as the title naming and set its "approved" status. The administrator names the title of the record 16 06 and checks its validity 16 08. If the title is invalid, the administrator will return to the management page presented in step 16 0 5 and provide the opportunity to name the record again. If the title is valid, the record recorded in the system database will be updated 16 1 0, and the approver will set the recorded “approval” attribute 1 6 1 4. If the recording is not approved 16 16, the approver will present a screen summarizing the changes made to the recording records in the database 1 6 1 8 and terminate the workflow 1 62 0. When the administrator approves the recording 1 6 1 6, the database will be updated to reflect the change in approval status 1 6 2 2 and then three actions will be performed simultaneously. Send a copy of the recording as an e-mail attachment to a specific archive custodian 1 6 2 6 After the administrator approves the recording, a day-to-day summary of the changes made to the recording in the database will appear 1 6 1 8 and then Perform the publisher 1 6 2 4 described later. The results of the publisher will be transmitted to a specific web server via FTP, and the web page will be part of it. After these three steps have been performed, the workflow will end 1620. The publisher is also a software subsystem initialized by DIVR to generate text data indicating that the location of the voice data is on a network storage device, a streaming server performing live broadcasting, or both. Just like workflow software, publishers include a series of directives like PERL or PYTHON

第28頁 五、發明說明(26) σ寫成的指令碼、一系别丨ν r / r丄丄μ ^ 程式,赤去4 : 土乐列以c/c + +语言寫成的組譯可執行 粗 或者k兩者的組合。經過工作流程的庳用之德,次 抖可就藉由電子郵件傳逆終佶 的應用之後貝 且嵌入網頁中。傳运給使用者或傳給網路伺服器,並 執:=本=一個具體實施例’呈現-般出版者所 者出版者一系列參數以執行該出版 复中-個依照出版者内部資料結構所設定1704。 ;:!系統資料庫17。8。不管有無取得其::數?= 耻Ρ曰攸接收的貪料中建立一個s〇l杳 % 在系統資料庫中執行1714 查詢 =進行格式化im,而該參數可依 =不同的…然後將格式化的結果傳回通話=果而 具有進階存取許可的使用者可檢視以及h ,、洗貝枓庫内含的資料,管理員可能會有受 &gt; 邊 J的存取特核,可對該公司的使用者、工法 f者以及排定的事件進行新增或編輯。而呈有較:二、出 ;^ ^ Μ (,.| superuser&quot; ) / ^ ^ 外工作流程、出版者以及職的事件進行新增用 此外,superuser還可新增和編輯祠服器、伺服器蓄扁輯。 公司0該網路式咨斗止t 4 最集和 覽出版者、iik —、枓庫丽^ t、應的其他功能允許管理員3 現牵涉到新增、編輯和丨劉覽的通用程式化;J:。圖18呈Page 28 V. Description of the invention (26) Instruction code written by σ, a series of 丨 ν r / r 丄 丄 μ ^ program, Chi Qu 4: Tuilei translation in c / c + + language executable Coarse or k. After the application of the workflow, the jitter can be transmitted via email and then embedded in the web page. To the user or to the web server, and execute: = 本 = a specific embodiment 'present-general publisher's publishers a series of parameters to perform the publishing re-invention-a structure based on the publisher's internal data Set to 1704. ;:! System Database 17.8. Whether or not to get its :: number? = Establish a s〇l 杳% in the greed received by shame P. Perform 1714 query in the system database = format im, and this parameter can be changed according to = different ... and then return the formatted result to the call = As a result, users with advanced access permissions can view and store the data contained in the database, and the administrator may have access to the &gt; side J ’s access auditing, which may allow users of the company, Add or edit work methods and scheduled events. And there are more: two, out; ^ ^ Μ (,. | Superuser &quot;) / ^ ^ external workflow, publisher and job events to add new addition In addition, superuser can also add and edit temple server, servo Device storage flat series. Company 0 The Internet-based consultation platform t 4 Most collections and publishers, iik —, 枓 Kuli ^ t, Ying and other features allow the administrator 3 is now involved in the addition, editing and general programming of Liu Lan ; J :. Figure 18 shows

第29頁 479429 五、發明說明(27) 在新增資料時,管理員首先選擇新增所要資料所需的人 適表單或範本1 8 02,例如:若管理員要新增新使用者到 統内,則應使用’’add user’,表單。在填完表單上的資 欄位後1 8 0 4,會檢查填入貧料的一致性丨8 〇 6,若資料鱼' 定樣板不符1 808,例如填入一個以上的編碼器位址,^合 顯示錯誤訊息1810。在此提供一個可讓使用者提出正二 單的機會1 8 1 2,就是再次檢查正確性丨8 〇 6。當填入的資 正確無誤1 8 0 8,該紀錄就會新增至資料庫内1 8 1 4。 &quot; 在編輯資料時,系統會從資料庫中取得「項目」表 單’犯例項目有使用者、公司或伺服器。然後管理員從: 回的項目清單中選取項目1818,然後使用者從選取的' &lt; 中選取兀件1819。例如’若管理員要編輯一位使用者,、 ,先從資料庫傳回的項目清單中選取,,users,,,然後管理 =再從所有使用者清單中選取特定使用者。系統從資料庫 T取得有關選取70件之資訊182(),並且將它放在可編 早中呈現給官理員1 822。經由表單對資料進行 、 ,後檢查確定新的資料與預定樣板一致1 826。若資料和預 疋樣本不一致1 8 28,將會顯示錯誤訊息丨83 〇並且提供管理 員一個修正錯誤的機會1 832,然後會再次檢查修正過資料 的一致性1 82 6。當資料格式正確時1 828,&amp; 資料寫回資料庫’藉以更新的紀錄1 834。 ^ 劉覽時,會從資料庫中取得項目清單1 836,管理員選取 一個項目1 83 8以及分在該項目底下的元件1 839。從資料庫 中將取得有關選取元件的資訊184〇並且執行動作丨842。產Page 29 479429 V. Description of the invention (27) When adding data, the administrator first selects the appropriate form or template for adding the required data 1 8 02, for example: if the administrator wants to add a new user to the system Inside, you should use `` add user '' form. After completing the information field on the form, 1 0 0 4 will check the consistency of the poor material. 8 0 6 If the data fish 'fixed template does not match 1 808, for example, more than one encoder address is entered, ^ The error message 1810 is displayed. Here is an opportunity for the user to submit a positive 2 order 1 8 1 2 is to check the correctness again. When the information entered is correct 1 8 0 8, the record will be added to the database 1 8 1 4. &quot; When editing the data, the system will get the "item" form from the database. The offending item is user, company or server. Then the administrator selects item 1818 from the list of returned items, and then the user selects component 1819 from the selected '&lt;. For example, ‘If the administrator wants to edit a user,, first select ,, users, from the list of items returned from the database, and then manage = and then select a specific user from the list of all users. The system obtains the information 182 () about 70 items from the database T, and presents it to the administrator 1 822 in an editable morning. The data were processed through the form, and then checked to confirm that the new data was consistent with the predetermined template 1 826. If the data and the pre-samples are inconsistent 1 8 28, an error message 丨 83 〇 will be displayed and the administrator will be given an opportunity to correct the error 1 832, and then the consistency of the corrected data will be checked again 1 82 6. When the format of the data is correct 1 828, &amp; the data is written back to the database ′ to update the record 1 834. ^ At the time of Liu Lan, the item list 1 836 was obtained from the database. The administrator selected an item 1 83 8 and the component 1 839 under the item. Information about the selected component is obtained from the database 184 and actions 842 are performed. Produce

479429 五、發明說明(28) 生的結果將顯示給管理員1 8 4 4,而管理員可透過編輯介面 修改元件。 雖然藉由較佳具體實施例來描述與說明本發明,但是精 通此技藝的人士可在不悖離本發明精神與領域之下進行許 多變更及修改,因此本發明並不受限於精確的方法學細節 或上述公佈的結構,所以本發明領域内包含許多變更與修 改。479429 V. Description of the invention (28) The results produced will be displayed to the administrator 1 8 4 4 and the administrator can modify the components through the editing interface. Although the present invention is described and illustrated by the preferred embodiments, those skilled in the art can make many changes and modifications without departing from the spirit and field of the present invention, so the present invention is not limited to precise methods The details of the study or the structure of the above publication, so many changes and modifications are included in the field of the present invention.

第31頁Page 31

Claims (1)

^/^429^ / ^ 429 資料之方法 言亥方法 • ~種經由電腦網路流通電話語音 包含: 、過來自語音資料作者的來電接收語音資料; 2將語音資料編碼成為目標檔案格式:以及 =、扁碼過的語音資料儲存在執立泣 了存取的檔案系統中。 …日抓軟肽之電腦」 2 ·如申請專利範圍第1項 資料到複數個編碼器之一。去,s亥方法包含繞送語音 3如中請專利範圍第2項之方法 音貢料作者的直好兮定莫人姑, 八甲依......且伴隨語Method of data: Speech method • ~ Voice of phone call through computer network includes:, Receive voice data through incoming call from the author of the voice data; 2 Encode the voice data into the target file format: and =, store the pasted voice data In the file system of the executive weeping access. … The computer that catches soft peptides every day "2 · For example, the data in the first item of patent application range to one of a plurality of encoders. Go, the Hai method includes routing voices. 3 The method of the second scope of the patent application is as follows. The author of the audio-visual materials is Ding Mo Rengu, Bajiayi ... and the accompanying words. 之-的步驟又疋導入繞送語音資料到複數個編瑪i 4.如申請專利範圍第1項之方法,言亥方法包含: 將一位置指定給編碼過的語音資料;以及 署值在!!語音資料儲存在檔案系統之前,將語音資料的、 置傳迗給官理員進行核准。 5·如申,專利範圍第丨項之方法,該方法包含依照一組 伴Ik δ吾曰貝料作者的規則,讓資料流語音用戶端可取得》 碼過的語音資料位置。The steps of-are to import the voice data to a plurality of editors. 4. If the method of the scope of patent application is the first, the method includes: assigning a position to the encoded voice data; and the value is in! ! Before the voice data is stored in the file system, transfer the voice data to the official for approval. 5. Rushen, the method of the first item of the patent scope, which includes a set of rules that accompany the author of Ik δ, so that the voice client of the data stream can obtain the position of the voice data that has been coded. .6·如申請專利範圍第1項之方法,其中由Real Audi〇TM壓 縮/解壓縮程式執行編碼步驟。 7.如中請專利範圍第1項之方法,其中由QuickTimeTM壓 縮/解壓縮程式執行編碼步驟。 8·如申請專利範圍第1項之方法,其中由wind〇ws MediaTM壓縮/解壓縮程式執行編碼步驟。.6. The method of claim 1 in which the encoding step is performed by a Real Audi ™ compression / decompression program. 7. The method according to item 1 of the patent, wherein the encoding step is performed by a QuickTimeTM compression / decompression program. 8. The method according to item 1 of the scope of patent application, wherein the encoding step is performed by a Windows MediaTM compression / decompression program. 第32頁 六、申請專利範圍 其中依照· w a v格式導 9 ·如申請專利範圍第〗項之方法 入編碼步驟。 〆 1 0 · —種經由電腦網路續 該系 統包含: ㈣^電話語音資料之系統 一第一電腦,其中右駐 的語音資料; 、σ以接收透過電話網路傳輪 一弟--電腦,其中古Pt 4·, 資料編碼成目的檔案袼式、、、广軟體可以直接將接收的語音 一資料儲存裝置,用 一第三電腦,其中右^存、,扁碼過的語音資料;以及 的軟體。 °以流傳所儲存編碼過語音資料 11 ·如申請專利範圊坌]Λ / 含複數個由複數個不同編、之糸,其中該第二電腦包 也 _ j、,、扁碼軟體構成之電腦;以及 第二電腦::腦’用於執行軟體將語音資料繞送到選取的 包含π ii作者的貧料將語音資料繞送到複數個 '3不同由編碼軟體的電腦之一(含)以上。 腦,該電腦依照即時/j1項之糸、统’該系統包含一電 到複數個包含不同紙本的作者喜好設定將語音資料繞送 14. 如申請專利範^軟體的'腦之一(含)以上。 含R e a 1 A u d i oTM壓始/第〇項之系統’其中該編碼軟體包 15. 如申請專=^堅縮程式。 第1 0項之系統,其中該編碼軟體包Page 32 6. Scope of patent application Among them, according to the w av format guide 9. If the method of the scope of the patent application item No. 〖enter the encoding step. 〆1 0 · —continued via a computer network. The system includes: ㈣ 电话 Phone voice data system-a first computer, of which the voice data resides on the right; σ to receive a younger brother via a phone network-computer, Among them, the ancient Pt 4 ·, the data is encoded into the destination file format, and the software can directly receive the received voice into a data storage device, using a third computer, of which the right and left are stored, and the coded voice data; and software. ° The coded voice data stored in circulation is used. 11 · If a patent application is filed] Λ / Contains a plurality of computers composed of a plurality of different codes, where the second computer package is also a computer composed of software And the second computer: "brain" is used to execute software to route voice data to a selected source containing the π ii author to route voice data to one of a plurality of '3 different computers (including) or more . Brain, the computer according to the real-time / j1 item, the system 'the system contains a power to a plurality of authors including different papers to set the voice data routing. )the above. System with Rea 1 A u d i oTM start / item ’where the coded software package 15. If applying for a special = ^ contraction program. The system of item 10, wherein the encoding software package 第33頁 479429 六、申請專利範圍 ^ 含Qu i c kT i meTM壓縮/解壓縮程式。 , 1 6.如申請專利範圍第1 0項之系統,其中該編碼軟體包 含W i n d 〇 w s M d e d i aTM壓縮/解壓縮程式。 1 7.如申請專利範圍第1 0項之系統,其中該編碼軟體包 含· wav格式。 1 8. —種經由電腦網路流通電話語音資料之系統,該系 統包含: 一部電腦,包含: 接收從電話網路傳輸來的語音資料之第一軟體; 直接將接收的語音資料編碼成目的檔案格式之第二^ 軟體; 流傳編碼過的語音資料之第三軟體;以及 一用於儲存已編碼語音資料的資料儲存裝置; 以及两電路連接至網路的電腦,該電腦可讓執行用 戶端軟體的用戶端電腦存取以接收資料流語音資料。 1 9. 一種經由電腦網路流通電話語音貢料之方法’該方 法包含: 由電腦接收打來的電話,該通電話是由語音資料作者 打來的; 根據作者取得個人描述資料; 根據作者個人描述檔内含的資料,將語音資料直接編 碼成目的檔案格式;以及 將編碼過的語音資料儲存在執行語音流軟體之電腦可 ’ 存取之檔案系統中。 ,Page 33 479429 6. Scope of patent application ^ Contains Qu i c kT i meTM compression / decompression program. 16. The system according to item 10 of the scope of patent application, wherein the encoding software includes a Winn d 0 w s M d e d i aTM compression / decompression program. 1 7. The system according to item 10 of the patent application scope, wherein the encoding software includes a · wav format. 1 8. —A system for distributing telephone voice data through a computer network, the system includes: a computer including: the first software for receiving voice data transmitted from the telephone network; directly encoding the received voice data into a purpose The second software in the file format; the third software that circulates the encoded voice data; and a data storage device for storing the encoded voice data; and a computer connected to the network by two circuits, which allows the client to run The software client computer accesses to receive streaming voice data. 1 9. A method for distributing telephone voice data via a computer network 'The method includes: receiving a call from a computer, the call is made by a voice data author; obtaining personal description data based on the author; Describe the data contained in the file, directly encode the voice data into the destination file format; and store the encoded voice data in a file system that can be accessed by a computer running voice streaming software. , 第34頁 479429 六、申請專利範圍 2 〇.如申請專利範圍第1 9項之方法,該方法包含: 根據使用者喜好設定描述檔内含的資料,將語音資料 繞送給複數個編碼器;以及 由複數個編碼器將語音資料編碼。 2 1.如申請專利範圍第2 0項之方法,其中語音資料的編 碼步驟幾乎是同時發生。 2 2.如申請專利範圍第1 9項之方法,其中個人描述檔為 依照XML法則格式化的資料。 2 3.如申請專利範圍第1 9項之方法,其中語音資料的編 碼步驟包含依照作者個人描述檔内含的資料設定編碼特 性。 2 4 ·如申請專利範圍第1 9項之方法,其中已編碼語音資 料的儲存步驟包含將已編碼語音資料儲存在作者個人描述 檔内指定之儲存裝置上。 2 5 ·如申請專利範圍第1 9項之方法,包含根據作者個人 描述檔執行工作流程。Page 34 479429 6. Application for Patent Scope 2 〇. The method for applying for Patent Scope Item 19, the method includes: setting the data contained in the description file according to user preferences, and routing the voice data to a plurality of encoders; And the speech data is encoded by a plurality of encoders. 2 1. The method according to item 20 of the scope of patent application, wherein the encoding steps of the voice data occur almost simultaneously. 2 2. The method according to item 19 of the scope of patent application, wherein the personal description file is data formatted in accordance with XML rules. 2 3. The method according to item 19 of the scope of patent application, wherein the encoding step of the voice data includes setting the encoding characteristics according to the data contained in the author's personal profile. 2 4 · The method of item 19 in the scope of patent application, wherein the step of storing the encoded voice data includes storing the encoded voice data on a storage device designated in the author's personal profile. 2 5 · The method of item 19 in the scope of patent application, including the execution of the workflow according to the author's personal profile. 第35頁Page 35
TW89119345A 1999-09-20 2000-09-20 System and method for distributing telephone audio data via a computer network TW479429B (en)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
US15476999P 1999-09-20 1999-09-20
US17983100P 2000-02-02 2000-02-02

Publications (1)

Publication Number Publication Date
TW479429B true TW479429B (en) 2002-03-11

Family

ID=26851765

Family Applications (1)

Application Number Title Priority Date Filing Date
TW89119345A TW479429B (en) 1999-09-20 2000-09-20 System and method for distributing telephone audio data via a computer network

Country Status (1)

Country Link
TW (1) TW479429B (en)

Similar Documents

Publication Publication Date Title
US7415537B1 (en) Conversational portal for providing conversational browsing and multimedia broadcast on demand
US7062709B2 (en) Method and apparatus for caching VoiceXML documents
US8700694B2 (en) Systems and methods for managing workflow based on multi-level specification of job processing requirements
US8326914B2 (en) Network system extensible by users
US7676473B2 (en) Propagation of user preferences to end devices
JP4597383B2 (en) Speech recognition method
US6507817B1 (en) Voice IP approval system using voice-enabled web based application server
US7167830B2 (en) Multimodal information services
US20030140121A1 (en) Method and apparatus for access to, and delivery of, multimedia information
US20020124100A1 (en) Method and apparatus for access to, and delivery of, multimedia information
US20070203927A1 (en) System and method for defining and inserting metadata attributes in files
JPH11512200A (en) Online / information service access and delivery system
US11451591B1 (en) Method and system for enabling a communication device to remotely execute an application
KR20080072641A (en) Streaming distribution of multimedia digital documents via a telecommunication network
TW479429B (en) System and method for distributing telephone audio data via a computer network
US20060165102A1 (en) Individual update or reorganisation of message and dialogue services specific to service providers
TW582153B (en) Method and system for providing real-time streaming services
US20030236666A1 (en) System for accessing a database using human speech
CA2352894C (en) Automatic electronic document processor system
CN106899637A (en) Internet accessing multi-language intelligent identifying system
WO2001022711A1 (en) System and method for distribution of telephone audio data via a computer network
US20040133654A1 (en) Method for adding sound via a telephone network to a page of data that can be remotely consulted via a communication network, site, voice server and computer using said method
KR20100058687A (en) Apparatus and method for processing speech recognition for large vocabulary speech recognition
JP2003271376A (en) Information providing system
WO2008009158A1 (en) A system and method for a multi-languages speech domain name and a voice search based on internet

Legal Events

Date Code Title Description
GD4A Issue of patent certificate for granted invention patent
MM4A Annulment or lapse of patent due to non-payment of fees