TW201240388A - System and method for video conference - Google Patents

System and method for video conference Download PDF

Info

Publication number
TW201240388A
TW201240388A TW100109026A TW100109026A TW201240388A TW 201240388 A TW201240388 A TW 201240388A TW 100109026 A TW100109026 A TW 100109026A TW 100109026 A TW100109026 A TW 100109026A TW 201240388 A TW201240388 A TW 201240388A
Authority
TW
Taiwan
Prior art keywords
audio
buffer
frequency
tone multi
signal
Prior art date
Application number
TW100109026A
Other languages
Chinese (zh)
Other versions
TWI448108B (en
Inventor
Chun-Yu Wang
Hsu-Cheng Lin
Chien-Hung Liu
Original Assignee
Ind Tech Res Inst
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 Ind Tech Res Inst filed Critical Ind Tech Res Inst
Priority to TW100109026A priority Critical patent/TWI448108B/en
Publication of TW201240388A publication Critical patent/TW201240388A/en
Application granted granted Critical
Publication of TWI448108B publication Critical patent/TWI448108B/en

Links

Abstract

A video conference system includes a phone, a phone adapter and a network communication device. The phone generates a first dual-tone multi-frequency (DTMF) signal. The phone adapter detects the first DTMF signal and according generates a first command string. The network communication device transforms the first command string into a first keyboard key action to control an application program according to a key translation profile. The network communication device receives a first audio signal form a network, and detects whether the first audio signal includes a second DTMF signal. If the first audio signal includes the second DTMF signal, the network communication device eliminates audio part corresponding to the second DTMF signal of the first audio signal according to detection results, and outputs the audio-eliminated first audio signal to the phone.

Description

201240388 i ττ /»» / τι r\ 六、發明說明: 【發明所屬之技術領域】 本發明是有關於一種視訊會議系統及方法。 【先前技術】 電話從發明至今是非常重要的通訊工具。傳統電話藉 由轉換收話端的電話號碼並發送一連串的雙音多頻 (dual-tone multi-frequency ’ DTMF)訊號給公共交換電話網 路(public switched telephone network ’ PSTN)的交換機,相 關的交換機透過DTMF訊號了解收話端的位置並建立連 線(call setup),之後兩端可以進行通話。其中連線建立的 基礎在於DTMF訊號,而電信服務商附加的各種服務亦是 在連線建立後’由使用者按下額外的數字按鍵,再由話機 送出對應的DTMF訊號給服務端進行轉換而得以提供對 應的服務。因此,DTMF訊號在整個電信網路中扮演了重 要的角色。 然而’隨著網際網路的蓬勃發展,網路電話(v〇Ip)逐 漸取代傳統的通話方式。亦即,連線建立的基礎已由轉換 收話端的電話號碼為DTMF訊號轉變為取得收話端的乡' 際網路通訊協定(IP)位址並透過網際網路而建立連、線' 網路電話或是通用序列匯排流(USB)電話轉接器所 而 具 傳統話機則演變成利用數字按鍵輸入IP垃址% ^ DTMF訊號則變成單純的數字訊號。 【發明内容】 201240388 1 / J /*»1 Γ\ 本揭露是有關於一種視訊會議系統及方法,利用將雙 音多頻訊號對應的字串命令轉換成鍵盤按鍵動作以控制 應用程式,JU貞測幻肖除音頻錢巾輕音多頻訊號,故 避免系統控制受到干擾而適用於視訊會議。 根據本揭露之第—方面’提出—種視訊會議系統,包 括-話機、-電話轉接器以及一網路通訊裝置。話機用以 產生-第-雙音多頻訊號。電話轉接器連接至話機,並用 以伯測第-雙音多頻訊號且據以產生一第一命令字串。網 路通訊裝置連接至電話轉接器與一網路,並用以依據一按 鍵轉換表將第一命令字串轉換為一第一鍵盤按鍵動作以 控2-應用程式。其中,網路通訊裳置更用以經由網路接 收第一音頻訊號,並偵測第一音頻訊號是否包括一第二 雙音多頻訊號。若第-音頻訊號包括第二雙音多頻訊號, 況裝置更用以依據偵測的結果將第—音頻訊號中 立f第f雙音多頻訊號之部分進行靜音處理,並用以將靜 «处理後之第-音頻訊號經由電話轉接器輸出至話機。 根據本揭露之第二方面’提出一種視訊會議方法,應 用於-視訊會議系統,視訊會議系統包括一 =器=網路通訊裝置,接器連接至話機,網路 列=、接至電5舌轉接器與網路。視訊會議方法包括下 S驟=用話機產生一第一雙音多頻訊號。利用電話轉 由。負測第—雙音多頻訊號且據以產生-第-命令字 =用網路通訊裝置依據一按鍵轉換表將第一命令字串 通訊鍵盤按鍵動作以控制一應用程式。利用網路 、左由網路接收-第一音頻訊號,並偵測第一音頻 4 201240388 说號疋否包括—楚―嫌也夕试 上 第二雙音乡若第—音頻訊號包每 馮讯唬,利用網路通訊裝置依據偵測的結果 第一音頻訊號中料1¾够 ^ j£u ^ Λ 、 、 & Τ對應第二雙音多頻訊號之部分進行靜屯 處理’並將靜音處理後之第一音頻訊號經由電 出至話機。 付牧裔輪 為了,本揭露之上述及其他方面有更佳的瞭解,下夂 特舉實知例,並配合所附圖式,作詳細說明如下: 【實施方式】 本揭露所提出之視訊會議系統及方法,利用將雙音多 頻讯唬對應的字串命令轉換成鍵盤按鍵動作以控制應用 程式,並偵測且消除音頻訊號中的雙音多頻訊號,故避免 系統控制受到干擾而適用於視訊會議。 請參照第1圖,其繪示依照一實施例之視訊會議系統 之方塊圖。視訊會議系統1〇〇包括一話機110、一電話轉 接器(phone adapter) 120以及一網路通訊裝置(netw〇rk communication device)130。電話轉接器120連接至話機 110;網路通訊裝置130連接至電話轉接器120與一網路(未 繪示於圖)。網路通訊裝置130例如為一主機(host)或一機 上盒(set-top box),其例如利用一通用序列匯排流(USB)與 電話轉接器120連接,然並不限制。網路通訊裝置130包 括一網路模組140以及一音頻模組(audio module)150。網 路模組140用以與網路建立通訊,故得以接收或傳送視訊 會議相關的音頻訊號及視頻訊號。網路通訊裝置130實質 上可接收來自話機U0之音頻訊號或是從網路接收遠端視201240388 i ττ /»» / τι r\ VI. Description of the Invention: [Technical Field of the Invention] The present invention relates to a video conferencing system and method. [Prior Art] The telephone has been a very important communication tool since its invention. The traditional telephone transmits the serial-tone multi-frequency ' DTMF signal to the switch of the public switched telephone network ' PSTN by switching the telephone number of the receiving end to the switch. The DTMF signal knows the location of the receiving end and establishes a call setup, after which the two ends can make a call. The basis for the establishment of the connection is the DTMF signal, and the various services attached by the telecommunication service provider are also after the connection is established. 'The user presses the extra numeric button, and then the corresponding DTMF signal is sent by the phone to the server for conversion. The corresponding service is available. Therefore, DTMF signals play an important role in the entire telecommunications network. However, with the rapid development of the Internet, Internet telephony (v〇Ip) has gradually replaced traditional calling methods. That is to say, the basis for the establishment of the connection has been changed from the telephone number of the converted receiving terminal to the DTMF signal to obtain the home network protocol (IP) address of the receiving end and establish a network connection through the Internet. A telephone or a universal serial stream (USB) telephone adapter has a conventional telephone that has evolved to use the digital keypad to input an IP address. The % DTMF signal becomes a simple digital signal. SUMMARY OF THE INVENTION 201240388 1 / J /*»1 Γ\ This disclosure relates to a video conferencing system and method for converting a string command corresponding to a dual tone multi-frequency signal into a keyboard button action to control an application, JU 贞In addition to the audio money towel, the audio-visual multi-frequency signal is used to avoid the interference of the system control and is suitable for video conferencing. According to a first aspect of the present disclosure, a video conferencing system includes a telephone, a telephone adapter, and a network communication device. The phone is used to generate a -Dual-tone multi-frequency signal. The telephone adapter is connected to the telephone and uses a beta-dual tone multi-frequency signal to generate a first command string. The network communication device is connected to the telephone adapter and a network, and is configured to convert the first command string into a first keyboard button action to control the 2-application according to a key conversion table. The network communication device is further configured to receive the first audio signal via the network, and detect whether the first audio signal includes a second dual tone multi-frequency signal. If the first audio signal includes the second dual tone multi-frequency signal, the device is further configured to mute the portion of the first audio signal neutral f-dual tone multi-frequency signal according to the detection result, and is used to process the static «processing The subsequent first-audio signal is output to the telephone via the telephone adapter. According to a second aspect of the present disclosure, a video conferencing method is applied to a video conferencing system, where the video conferencing system includes a device=network communication device, the connector is connected to the phone, the network column=, and the battery is connected to the battery. Adapter and network. The video conferencing method includes the following steps: generating a first dual tone multi-frequency signal by using the telephone. Use the phone to transfer. The first-dual-tone multi-frequency signal is negatively measured and the -first-command word is generated. The first command string communication keyboard button is operated by the network communication device according to a key conversion table to control an application. Use the network, left to receive the first audio signal from the network, and detect the first audio 4 201240388 No. 疋 No Include - Chu 嫌 也 试 试 第二 第二 第二 第二 第二 第二 第二 — — — — — — — — — —唬, using the network communication device according to the result of the detection, the first audio signal in the first audio signal is sufficient, and the portion corresponding to the second dual-tone multi-frequency signal is subjected to the quiet processing 'and the mute processing The first audio signal is then sent out to the phone. In order to better understand the above and other aspects of the disclosure, the following is a detailed description of the examples, and the detailed description is as follows: [Embodiment] The video conference proposed by the disclosure The system and method use the string command corresponding to the dual tone multi-frequency signal to convert the keyboard key action to control the application, and detect and eliminate the dual tone multi-frequency signal in the audio signal, thereby avoiding the system control being interfered with. In the video conference. Please refer to FIG. 1 , which is a block diagram of a video conferencing system in accordance with an embodiment. The video conferencing system 1 includes a telephone 110, a telephone adapter 120, and a netw〇rk communication device 130. The telephone adapter 120 is coupled to the telephone 110; the network communication device 130 is coupled to the telephone adapter 120 and a network (not shown). The network communication device 130 is, for example, a host or a set-top box, which is connected to the telephone adapter 120, for example, using a universal serial stream (USB), but is not limited. The network communication device 130 includes a network module 140 and an audio module 150. The network module 140 is used to establish communication with the network, so that it can receive or transmit audio signals and video signals related to the video conference. The network communication device 130 can substantially receive the audio signal from the phone U0 or receive the far-end view from the network.

201240388 ' A 里 w /*> /ηγλ sfl會5義相關的音頻訊號。 音頻模組 150 包括一雙音多頻(dual-tone multi-frequency ’ DTMF)訊號轉發(translation)模組 162、 一鍵盤事件發射器(keyboard event launcher) 164、一雙音多 頻消除(elimination)模組166、一音頻輸出模組168以及一 音頻輸入模組169。音頻輸出模組168例如為一耳機,音 頻輸入模組169為本機所接之實體裝置例如為一麥克風, 音頻輸入模組169可接收來自網路的音頻封包而得到音頻 訊號。於本實施例中,當視訊會議的連線建立後,話機11〇 可作為控制器使用以透過電話轉接器120與網路通訊裝置 130控制本地端的應用程式。請同時參照第2A圖及第2B 圖’其分別繪示依照一實施例之視訊會議方法之部分流程 圖。其中,分別繪示於第2A圖及第2B圖之視訊會議方法 可同時進行’亦可以互為連續的行為,並不限制。於第2A 圖之步驟S200中,按壓話機110的數字按鍵而產生一第 雙音多頻訊號。於步驟S210中,電話轉接器12〇偵測 第一雙音多頻訊號且據以產生一第一命令字串。 於步驟S220中,利用雙音多頻訊號轉發模組162從 電話轉接器接收120第一命令字串並將第一命令字串轉發 至鏠盤事件發射器164。於步驟S230中,鍵盤事件發射器 164依據一按鍵轉換表將第一命令字串轉換為第一鍵盤按 鍵動作’本地端之一應用程式在收到第一鍵盤按鍵動作後 即會執行對應的處理 工作而改變使用者介面,故話機110 可作為控制器控制本地端的應用程式。請參照第3圖,其 、9示依照一實施例之按鍵轉換表之示意圖。於第3圖中, 6 201240388 1 VV / J /ΗΓ/Λ 不同的話機數字按鍵對應至不同的鍵盤按鍵動作,例如話 機數字按鍵「01」對應至鍵盤動作「Tab」,話機數字按鍵 〇8」對應至鍵盤動作「D〇Wn」等。此外亦可以將單一話 機數字按鍵對應至組合鍵動作,但不限制,端視設計需求 而定。 當視訊會議進行中時,若遠端的使用者同時進行通話 及利用話機控制操作系統,則本地端的視訊會議系統iOO 在從網路接收視訊會議相關的視頻訊號與音頻訊號時,亦 會聽到遠端的按鍵音輸出’亦即網路模組14〇從網路接收 的音頻訊號會包括遠端送出的雙音多頻訊號,可能會導致 電話轉接器120產生誤判而使得網路通訊裝置丨30產生錯 誤控制本地端應用程式的情形發生。因此,於第2B圖之 步驟S240中,網路通訊裝置13〇利用網路模組140經由 網路接收一第一音頻訊號,此第一音頻訊號為來自遠端的 視訊會議相關音頻訊號。第一音頻訊號在傳送前會被以脈 碼調變(pulse code modulation,PCM)進行取樣,然後被壓 切割為多個第一音頻封包以進行傳送,其中每秒脈碼取 樣頻率例如為8000Hz。網路通訊裝置13〇在接收多個第一 音頻封包後會進行解析及解壓縮而得到脈碼調變聲音片 ^ 因此第一音頻sfl號包括多個第一音頻片段。 九之後,於步驟S25〇中,音頰輪入模組169偵測第一 曰頻成號是否包括一第二雙音多賴訊號。音頻輸入模組 =9實際上會先偵測每個第一音頰片段之一 d τ M F旗標, 右此旗標被設定為偽值(False),則此第一音頻片段不需被 偵測而可直接被播放;若此旗標被設定為真值(Tr u e),則 201240388 >201240388 ' A w / * > / η γ λ sfl will be 5 sense related audio signals. The audio module 150 includes a dual-tone multi-frequency 'DTMF' signal translation module 162, a keyboard event launcher 164, and a dual tone multi-frequency elimination (elimination). The module 166, an audio output module 168 and an audio input module 169. The audio output module 168 is, for example, a headphone, and the audio input module 169 is a physical device connected to the device, for example, a microphone. The audio input module 169 can receive an audio packet from the network to obtain an audio signal. In this embodiment, after the connection of the video conference is established, the telephone 11 can be used as a controller to control the local application through the telephone adapter 120 and the network communication device 130. Referring to FIG. 2A and FIG. 2B, a partial flow diagram of a video conference method according to an embodiment is shown. The video conferencing methods shown in Figures 2A and 2B, respectively, can be performed simultaneously or in parallel, and are not limited. In step S200 of Fig. 2A, the digital button of the telephone 110 is pressed to generate a first dual tone multi-frequency signal. In step S210, the telephone adapter 12 detects the first dual tone multi-frequency signal and generates a first command string accordingly. In step S220, the dual command signal forwarding module 162 receives 120 the first command string from the telephone adapter and forwards the first command string to the disk event transmitter 164. In step S230, the keyboard event transmitter 164 converts the first command string into the first keyboard button according to a button conversion table. 'One of the local end applications will execute the corresponding processing after receiving the first keyboard button action. The user interface is changed by the work, so the phone 110 can be used as a controller to control the local application. Please refer to FIG. 3, which shows a schematic diagram of a key conversion table according to an embodiment. In Figure 3, 6 201240388 1 VV / J /ΗΓ/Λ Different phone number buttons correspond to different keyboard button actions, for example, the phone number button "01" corresponds to the keyboard action "Tab", the phone number button 〇 8" Corresponds to the keyboard action "D〇Wn" and so on. In addition, a single phone number button can be assigned to the combination button action, but it is not limited, depending on the design requirements. When a video conference is in progress, if the remote user simultaneously makes a call and uses the phone to control the operating system, the local videoconferencing system iOO will also hear the video signal and audio signal related to the video conference when receiving the video conference from the network. The button audio output of the terminal, that is, the audio signal received by the network module 14 from the network may include the dual-tone multi-frequency signal sent by the remote end, which may cause the telephone adapter 120 to misjudge the network communication device. 30 The situation that caused an error to control the local application occurred. Therefore, in step S240 of FIG. 2B, the network communication device 13 receives a first audio signal via the network using the network module 140, and the first audio signal is a video conference related audio signal from the far end. The first audio signal is sampled by pulse code modulation (PCM) before being transmitted, and then compressed into a plurality of first audio packets for transmission, wherein the pulse code sampling frequency per second is, for example, 8000 Hz. The network communication device 13 解析 parses and decompresses the plurality of first audio packets to obtain a pulse code modulated sound film. Therefore, the first audio sfl number includes a plurality of first audio segments. After the ninth step, in step S25, the buccal wheeling module 169 detects whether the first cymbal number includes a second doubling signal. The audio input module=9 will actually detect one d τ MF flag of each first buccal segment first, and the right flag is set to a false value (False), then the first audio segment does not need to be detected. The test can be played directly; if the flag is set to true (True), then 201240388 >

1 W7374PA 此第一音頻片段需被偵測後才能播放。音頻輸入模組169 具有一緩衝器(buffer),其大小例如為i〇k〜64k位元組 (byte)。 音頻輸入模組 169 具有一第一緩衝器 TMP_LINKED_LIST及一第二緩衝器Buffer。請參照第4 圖’其繪示依照一實施例之音頻輸入模組之第一緩衝器及 第二緩衝器之示意圖。於第4圖中,旗標被設定為真值的 多個第一音頻片段(例如α 〇〜α 3)被依序複製並暫存在第 一緩衝器 TMP_LINKED_LIST。當第一緩衝器 TMP_LINKED_LIST内之多個第一音頻片段之資料長度大 於一最大偵測長度MaxDetectionLength,第一緩衝器 TMP_LINKED_LIST内之多個第一音頻片段會被送到第二 緩衝器Buffer内合併並得到對應之一位移值Offset,此位 移值Offset為第一緩衝器TMP_LINKED_LIST内之多個第 一音頻片段之資料長度和最大偵測長度 MaxDetectionLength 相除之餘數。 音頻輸入模組169配合位移值Offset偵測第二緩衝器 Buffer内之多個第一音頻片段是否包括第二雙音多頻訊 號。亦即,音頻輸入模組169會偵測第二緩衝器Buffer内 對應位置0到位置MaxDetectionLength的内容,並額外/(貞 測第二緩衝器Buffer内對應位置Offset到位置 (MaxDetectionLength+Offset)的内容。此外,音頻輸入模組 169更反覆偵測第二緩衝器Buffer内之多個第一音頻片段 是否包括第二雙音多頻訊號,且反覆偵測的次數K實質上 可由話機110之一每秒脈碼調變取樣頻率(例如為 8 201240388 l W / J /ΗΓ/\ 8000Hz)、最大彳貞測長度MaxDetectionLength及一延遲範 圍常數DelayConstant所決定,例如下述之公式(1),其中 可藉由調整延遲範圍常數DelayConstant而得到最佳化之 反覆偵測的次數K。 K=ceil(8000/MaxDetectionLength)/Delay Constant (1) 若音頻輸入模組169偵測到第二緩衝器Buffer内之多 個第一音頻片段包括第二雙音多頻訊號,則於步驟S260 中,雙音多頻消除模組166將第一音頻訊號中對應第二雙 音多頻訊號之部分進行靜音處理。雙音多頻消除模阻166 實質上將第二緩衝器Buffer内之多個第一音頻片段設定為 〇值,亦即第一音頻片段α 〇〜《3均被填入0值。其中,被 設定為0值之多個第一音頻片段的時間長度SkipCount實 質上可由一延遲基本單位DelayEquivalent、話機110之一 每秒脈碼調變取樣頻率(例如為8000Hz)及最大偵測長度 MaxDetectionLength所決定,例如下述之公式(2)。 SkipCount=ceil(8000/MaxDetectionLength)x DelayEquivalent (2)。 接續於步驟S260之後,於步驟S270中,被雙音多频 消除模組16 6靜音處理後之第一音頻訊號輸出至音頻輪出 模組168,或者被雙音多頻消除模組166靜音處理後之第 一音頻訊號經由電話轉接器120輸出至話機110播敌。而 在步驟S250中,若音頻輸入模組169未偵測到第二、緩衝 器Buffer内之多個第一音頻片段包括第二雙音多頻訊藏, 則於步驟S280中,此些第一音頻片段會被直接播玫。 上述之步驟S240〜S280係用以當視訊會議進行中時 201240388 ’ 1 vv / j /*+r/\ 供本地端的視訊會議系統100濾除從遠端接收的視訊會議 相關的音頻訊號中的按鍵音輸出(亦即雙音多頻訊號),以 避免網路通訊裝置130產生錯誤控制本地端應用程式的情 形發生。更進一步地,在本地端的視訊會議系統100輸出 視訊會議相關的音頻訊號時,亦可先濾除音頻訊號中的按 鍵音輸出以避免遠端產生誤動作。 於步驟S300中,音頻輸入模組169或話機110產生 一第二音頻訊號。於步驟S310中,音頻輸入模組169偵 測第二音頻訊號是否包括一第三雙音多頻訊號。若第二音 頻訊號包括第三雙音多頻訊號,則於步驟S320中,雙音 多頻消除模組166依據電話轉接器120偵測的結果將第二 音頻訊號中對應第三雙音多頻訊號之部分進行靜音處 理,並將靜音處理後之第二音頻訊號經由網路模組140輸 出至網路。其中,音頻輸入模組169偵測雙音多頻訊號與 雙音多頻消除模組166消除雙音多頻訊號的原理已詳述於 步驟S250及S260的相關内容中,故於此不再重述。若第 二音頻訊號未包括第三雙音多頻訊號,則於步驟S330中, 直接將第二音頻訊號經由網路模組140輸出至網路。如此 一來,本地端的視訊會議系統100即可輸出不包括按鍵音 輸出(亦即雙音多頻訊號)的視訊會議相關的音頻訊號時, 遠端的收話端即不會產生誤判而錯誤控制應用程式的情 形。 本揭露上述實施例所揭露之視訊會議系統及方法,依 據電話轉接器接收的雙音多頻訊號產生對應的字串命 令,再將字串命令依據一按鍵轉換表轉換成鍵盤按鍵動作 201240388 以控制應用程式’利於進行視訊會議。更進一步地,本揭 露之視訊會議系統及方法彳貞測且消除從網路接收的音頻 訊號及欲傳送至網路的音頻訊號中的雙音多頻訊號,故可 避免本地端及遠端的視訊會議系統避免因誤判雙音多頻 sfL 而產生錯块控制應用程式的情形。 細上所述,雖然本發明已以多個實施例揭露如上,然 其並非用以限定本發明。本發明所屬技術領域中具有通常 知識者,在不脫離本發明之精神和範圍内,當可作各種之 更動與潤飾。因此’本發明之保護範圍當視後附之 利範圍所界定者為準。 201240388 l W /J /ΗΓΙ\ 【圖式簡單說明】 第1圖繪示依照一實施例之視訊會議系統之方塊圖。 第2Α圖及第2Β圖分別繪示依照一實施例之視訊會 議方法之部分流程圖。 第3圖繪示依照一實施例之按鍵轉換表之示意圖。 第4圖繪示依照一實施例之音頻輸入模組之第一緩 衝器及第二緩衝器之示意圖。 【主要元件符號說明】 100 : 視訊會議系統 110 : 話機 120 : 電話轉接器 130 : 網路通訊裝置 140 : 網路模組 150 : 音頻模組 162 : 雙音多頻訊號轉發模組 164 : 鍵盤事件發射器 166 : 雙音多頻消除模組 168 : 音頻輸出模組 169 : 音頻輸入模組 TMP_LINKED_LIST :第一緩衝器 Buffer :第二缓衝器 121 W7374PA This first audio clip needs to be detected before it can be played. The audio input module 169 has a buffer of a size of, for example, i〇k to 64k bytes. The audio input module 169 has a first buffer TMP_LINKED_LIST and a second buffer Buffer. Referring to FIG. 4, a schematic diagram of a first buffer and a second buffer of an audio input module according to an embodiment is shown. In Fig. 4, a plurality of first audio segments (e.g., α 〇 〜 α 3) whose flags are set to true values are sequentially copied and temporarily stored in the first buffer TMP_LINKED_LIST. When the data length of the plurality of first audio segments in the first buffer TMP_LINKED_LIST is greater than a maximum detection length MaxDetectionLength, the plurality of first audio segments in the first buffer TMP_LINKED_LIST are sent to the second buffer Buffer and merged. Corresponding to a displacement value Offset, the displacement value Offset is a remainder of the data length of the plurality of first audio segments in the first buffer TMP_LINKED_LIST and the maximum detection length MaxDetectionLength. The audio input module 169 cooperates with the displacement value Offset to detect whether the plurality of first audio segments in the second buffer Buffer include the second dual tone multi-frequency signal. That is, the audio input module 169 detects the content of the corresponding position 0 to the position MaxDetectionLength in the second buffer Buffer, and additionally / (measures the content of the corresponding position Offset to the position (MaxDetectionLength+Offset) in the second buffer Buffer In addition, the audio input module 169 further detects whether the plurality of first audio segments in the second buffer Buffer include the second dual tone multi-frequency signal, and the number of times of the repeated detection K is substantially one of each of the phones 110. The second pulse code modulation sampling frequency (for example, 8 201240388 l W / J / ΗΓ / 8000 Hz), the maximum detection length MaxDetectionLength and a delay range constant DelayConstant, for example, the following formula (1), which can be borrowed The number K of times of the overridden detection that is optimized by adjusting the delay range constant DelayConstant K=ceil(8000/MaxDetectionLength)/Delay Constant (1) If the audio input module 169 detects the number of buffers in the second buffer The first audio segment includes the second dual tone multi-frequency signal, and in step S260, the dual tone multi-frequency cancellation module 166 corresponds to the second audio-tone multi-frequency signal in the first audio signal. Performing a mute process. The dual tone multi-frequency cancellation mode 166 substantially sets a plurality of first audio segments in the second buffer Buffer to a threshold value, that is, the first audio segment α 〇 ~ "3 are filled with 0 value The time length SkipCount of the plurality of first audio segments set to a value of 0 may be substantially a delay basic unit DelayEquivalent, one of the telephones 110 per second pulse code modulation sampling frequency (for example, 8000 Hz) and a maximum detection length. MaxDetectionLength determines, for example, the following formula (2): SkipCount=ceil(8000/MaxDetectionLength)x DelayEquivalent (2). Following step S260, in step S270, it is muted by the dual-tone multi-frequency cancellation module 16 6 The first audio signal is output to the audio wheeling module 168, or the first audio signal silenced by the dual-tone multi-frequency cancellation module 166 is output to the phone 110 via the phone adapter 120. In step S250 If the audio input module 169 does not detect that the plurality of first audio segments in the buffer buffer include the second dual tone multi-frequency information, then in step S280, the first audio segments are The above steps S240~S280 are used when the video conference is in progress 201240388 '1 vv / j /*+r/\ for the local videoconferencing system 100 to filter out the video conference received from the far end. The key tone output (ie, the dual tone multi-frequency signal) in the audio signal prevents the network communication device 130 from generating an error to control the local end application. Further, when the videoconferencing system 100 of the local end outputs the audio signal related to the video conference, the button sound output in the audio signal may be filtered out first to prevent the remote terminal from malfunctioning. In step S300, the audio input module 169 or the phone 110 generates a second audio signal. In step S310, the audio input module 169 detects whether the second audio signal includes a third dual tone multi-frequency signal. If the second audio signal includes the third dual tone multi-frequency signal, then in step S320, the dual tone multi-frequency cancellation module 166 converts the third audio to the third audio signal according to the result detected by the phone adapter 120. The part of the frequency signal is muted, and the second audio signal after the mute processing is output to the network via the network module 140. The principle that the audio input module 169 detects the dual-tone multi-frequency signal and the dual-tone multi-frequency cancellation module 166 to eliminate the dual-tone multi-frequency signal is detailed in the related contents of steps S250 and S260, so it is not heavy here. Said. If the second audio signal does not include the third dual tone multi-frequency signal, then in step S330, the second audio signal is directly output to the network via the network module 140. In this way, when the local videoconferencing system 100 can output the audio signal related to the video conference that does not include the button sound output (that is, the dual tone multi-frequency signal), the remote receiving terminal does not generate false positives and is erroneously controlled. The case of the application. The video conference system and method disclosed in the above embodiments generate a corresponding string command according to the dual tone multi-frequency signal received by the telephone adapter, and then convert the string command into a keyboard button action 201240388 according to a button conversion table. The control application 'is convenient for video conferencing. Further, the video conferencing system and method of the present disclosure detect and eliminate the audio signal received from the network and the dual tone multi-frequency signal in the audio signal to be transmitted to the network, thereby avoiding the local end and the far end. The video conferencing system avoids the situation of a wrong block control application due to misjudgment of the dual tone multi-frequency sfL. The invention has been described above in terms of several embodiments, which are not intended to limit the invention. Those skilled in the art can make various changes and modifications without departing from the spirit and scope of the invention. Therefore, the scope of the invention is defined by the scope of the appended claims. 201240388 l W /J /ΗΓΙ\ [Simple Description of the Drawings] FIG. 1 is a block diagram of a video conferencing system according to an embodiment. 2 and 2 are respectively a partial flow chart of a video conferencing method according to an embodiment. FIG. 3 is a schematic diagram of a key conversion table according to an embodiment. 4 is a schematic diagram of a first buffer and a second buffer of an audio input module according to an embodiment. [Main component symbol description] 100 : Videoconferencing system 110 : Telephone 120 : Telephone adapter 130 : Network communication device 140 : Network module 150 : Audio module 162 : Dual tone multi - frequency signal forwarding module 164 : Keyboard Event emitter 166 : Dual tone multi-frequency cancellation module 168 : Audio output module 169 : Audio input module TMP_LINKED_LIST : First buffer Buffer : Second buffer 12

Claims (1)

201240388 1 vy I j /*tr/-% 七、申請專利範圍: 1 · 一種視訊會議系統,包括: 一話機,用以產生一第一雙音多頻訊號; 一電話轉接器,連接至該話機,並用以偵測該第一雙 曰夕頻sfl说且據以產生一第^一命令字串;以及 一網路通訊裝置,連接至該電話轉接器與一網路,並 用以依據一按鍵轉換表將該第一命令字串轉換為一第一 鍵盤按鍵動作以控制一應用程式; 其中,§亥網路通訊裝置更用以經由該網路接收一第一 音頻訊號,並偵測該第一音頻訊號是否包括一第二雙音多 頻訊號,若該第一音頻訊號包括該第二雙音多頻訊號,該 網路通訊裝置更用以依據偵測的結果將該第一音頻訊號 中對應該第二雙音多頻訊號之部分進行靜音處理,並用以 將靜音處理後之該第一音頻訊號經由該電話轉接器輸出 至該話機。 2·如申請專利範圍第1項所述之視訊會議系統,其 中該網路通訊裝置包括: 一網路模組’用以與該網路建立通訊;以及 一音頻模組,包括: 一鍵盤事件發射器,用以依據該按鍵轉換表將該第 一命令字串轉換為該第一鍵盤按鍵動作; 一雙音多頻訊號轉發模組,用以從該電話轉接器接 收該第一命令字串並將該第一命令字串轉發至該鍵盤事 件發射器; 一音頻輸入模組,用以經由該,網路接收該第一音頻 13 201240388 i w a ' 》 ^虎JU貞測4第—音頻訊號是否包括該第 頻訊 號; “势雙音多頻消除模组,用以將該第一音頻訊號中對 第二雙音多頻訊號之部分進行靜音處理;以及 一音頻輸出模組’用以將靜音處理後之該第一音頻 訊號經由該電話轉接器輸出至該話機。 4專利範圍帛2項所述之減會議系統,其 中該第-音頻訊號包括複數個第—音頻片段,該音頻輸入 模組具有一第一緩衝器及一第二緩衝器,該第一緩衝器用 以依序複製並暫存該些第一音頻片段,當該第一緩衝器内 之該些第-音頻片段之資料長度大於一最大偵測長度,該 第一緩衝器内之該些第—音頻片段於該第二緩衝器内被 合併並得到對應之一位移值,該音頻輸入模組配合該位移 值偵測該第二緩衝器内之該些第—音頻片段是否包括該 第-雙s夕頻3fL號’若該些第—音頻片段包括該第二雙音 多頻訊號,該雙音多頻消除模組將該第二緩衝器内之該些 第一音頻片段設定為〇值。 4.如申凊專利範圍第3項所述之視訊會議系統,其 t該音頻輸人模組更反覆_該第二緩衝器内之該些第 一音頻片段是否包括該第二雙音多頻_,且反覆制的 次數由該話機之-每秒脈碼調變取樣頻率、該最大侧長 度及一延遲範圍常數所決定。 5.如申凊專利範圍第3項所述之視訊會議系統,其 中被設定為0值之該些第一音頻片段的時間長度由^延遲 基本單位、該話機之一每秒脈碼調變取樣頻率及該最大偵 201240388 測長度所決定。 6·如申請專利範圍第2項所述之視訊會議系統,其 中泫活機更用以產生一第二音頻訊號,該音頻輸入模組更 用以伯測該第二音頻訊號是否包括一第三雙音多頻訊 號,右該第二音頻訊號包括該第三雙音多頻訊號,該雙音 夕頻消除模組更用以依據該音頻輸入模組偵測的結果將 =第二音頻訊號中對應該第三雙音多頻訊號之部分進行 靜s處理,並用以將靜音處理後之該第二音頻訊號經由該 網路模組輪出至該網路。 .如申凊專利範圍第6項所述之視訊會議系統,其 中該第二音頻訊號包括複數個第二音頻片段,該該音頻輸 模且八有帛、緩衝器及一第二緩衝器,該第一緩衝器 用以依序複製並暫存該些第二音頻片段,#該第—緩衝器 内之該些第二音頻片段之資料長度大於—最大债測^ 度,該第-緩衝器内之該些第二音頻片段於該第二緩衝器 内被合併並得騎應之—位移值,該音頻輸人模組配合該 位移值制該第二緩衝器内之該些第二音頻片段是 括^第王雙音錢喊,若該些第二音頻諸包括該第三 雙音多頻訊號’該雙音㈣消除模組將該第 該些第二音頻片段設定為〇值。 11益内之 8. Μ請專利範圍第7項所述之視訊㈣系統,並 中該曰頻輸入模組更反覆價測該第二緩衝器内之該此第 二音頻片段是否包括該第三雙音多頻— — 叉θ夕須況琥,且反覆偵測的 -人數由該箱之-母秒脈仙變取樣鮮、 度及一延遲範圍常數所決定。 偵測長 15 201240388 1 W/j/4^A 9·如申請專利範圍第7項所述之視訊會議系統,其 中被設定為0值之該些第二音頻片段的時間長度由一延遲 基本單位、該話機之一每秒脈碼調變取樣頻率及該最大偵 測長度所決定。 、 一種視訊會議方法,應用於一視訊會議系統,該 視訊會議系統包括一話機、一電話轉接器及一網路通訊裝 置,该電話轉接器連接至該話機,該網路通訊裝置連接至 該電話轉接器與一網路,該視訊會議方法包括: 利用該話機產生一第一雙音多頻訊號; 利用該電話轉接器偵測該第一雙音多頻訊號 產生一第一命令字串 八〜利用該網路通訊裝置依據一按鍵轉換表將該第一^ 令子串轉換為—第—鍵盤按鍵動作以控制—應用程式; 〇 利用該網路通訊裝置經由該網路接收一第一音頻言 :;並偵測該第一音頻訊號是否包括一第二雙音多頻言 網路號包括該第二雙音多頻訊號’利用該 节第二镂二、據彳貞測的結果將該第—音頻訊號中對應 二頻訊號之部分進行靜音處理,並將靜音處理 s頻訊號經由該電話轉接器輸出至該話機。 I中該網專利範圍帛1G項所狀視訊會議方法, 路模::用以=展置包括一網路模組及一音頻模組,該網 件發射器網路建立通訊,該音頻模組包括—鍵盤事 一雙音多頻訊賴發额、—音赌入模組、 承模組及一音頻輸出模組,該視訊會議方法 201240388 1 f ^ Ι-ΤΙ Γ\ 更包括: 利用該雙音多頻訊號轉發模組從該電話轉接器接收 該第一命令字串並將該第一命令字串轉發至該鍵盤事件 發射器; -利用該鍵盤事件發射器依據該按鍵轉換表將該第一 命令字串轉換為該第一鍵盤按鍵動作; 利用該音頻輸入模組經由該網路接收該第一音頻訊 號,並偵測該第一音頻訊號是否包括該第二雙音多頻訊 號; 4 *利用該雙音多頻消除模組將該第一音頻訊號中對應 §亥第二雙音多頻訊號之部分進行靜音處理;以及 利用"亥曰頻輸出模組將靜音處理後之該第一音頻訊 號經由該電輯接諸出至紐機。 1 /2.如申請專利範圍第u項所述之視訊會議方法, 其中該第一音頻訊號包括複數個第一音頻片段,該音頻輸 入模組具有—第—緩衝器及—第二緩衝器該視 法更包括: A·利用第一該缓衝器依序複製並暫存該些第一音頻片 段; 當該第-緩衝器内之該些第—音頻片段之#料長度 最大偵測長度’利用該第二緩衝器合併該第—緩衝 之该些第一音頻片段並得到對應之一位移值; 利㈣音頻輸人模組配合該位移幻貞測該第 :内:Γ第一音頻片段是否包括該第二雙音多頻訊 17 201240388 ., 1 W /ΗΓΛ 右该些第一音頻片段包括該第二雙音多頻訊號,利用 该雙音多頻消除模組將該第二緩衝器内之該些第一音頻 片段設定為〇值。 13. 如申請專利範圍第12項所述之視訊會議方法, 更包括: 利用該音頻輸入模組反覆偵測該第二緩衝器内之該 些第一音頻片段是否包括該第二雙音多頻訊號,且反覆偵 測的-人數由该話機之一每秒脈碼調變取樣頻率、該最大偵 測長度及一延遲範圍常數所決定。 14. 如申請專利範圍第12項所述之視訊會議方法, 其中被設定為〇值之該些第一音頻片段的時間長度由一延 遲基本單位、該話機之一每秒脈碼調變取樣頻率及該最大 偵測長度所決定。 15. 如申請專利範圍第11項所述之視訊會議方法, 更包括: 利用該音頻輸入模組或該話機產生一第二音頻訊號; 利用該音頻輸入模組偵測該第二音頻訊號是否包括 一第三雙音多頻訊號;以及 右該第二音頻訊號包括該第三雙音多頻訊號,利用該 雙音多頻消除模組依據該音頻輸人模組偵測的結果將該 第二音頻訊號中對應該第三雙音多頻訊號之部分進行靜 =處理,並心將靜音處理後之該第二音賴號經由該網 路模組輸出至該網路。 16·如申請專利範圍第15項所述之視訊會議方法, 其中該第二音頻訊號包括複數個第二音頻片段,該音頻輸 201240388 入模組具有一第一緩衝器及一第二緩衝器,該視訊會議方 法更包括: 利用該第一緩衝器依序複製並暫存該些第二音頻片 段; 當該第一緩衝器内之該些第二音頻片段之資料長度 大於一最大偵測長度,利用該第二緩衝器合併該第一緩= 器内之該些第二音頻片段並得到對應之一位移值; 利用該音頻輸入模組配合該位移值偵測該第二緩衝 益内之該些第二音頻片段是否包括該第三雙音多頻訊 號;以及 右β玄些第一音頻片段包括該第三雙音多頻訊號,利用 該雙音多頻消除模組將該第二緩衝器内之該些第二音頻 片段設定為〇值。 17.如申4專利feu第16項所述之視訊會議方法, ^中該音頻輸人模組更反覆偵測該第二緩衝器内之該些 ϊγΓ頻片段是否包括該第三雙音多頻訊號,域覆僧測 該話機之—每秒脈瑪難取樣鮮、該最大偵測 長度及一延遲範圍常數所決定。 並二申料㈣_16韻紅視訊會議方法, 遲美本::為0值之忒些第二音頻片段的時間長度由-延 話機之—每秒脈碼調變取樣頻率及該最大201240388 1 vy I j /*tr/-% VII. Patent application scope: 1 · A video conferencing system, comprising: a telephone for generating a first dual tone multi-frequency signal; a telephone adapter connected to the a phone, and configured to detect the first pair of sf sfl and generate a first command string; and a network communication device connected to the phone adapter and a network, and used to The key conversion table converts the first command string into a first keyboard button action to control an application; wherein the network communication device is further configured to receive a first audio signal via the network, and detect the Whether the first audio signal includes a second dual tone multi-frequency signal, and if the first audio signal includes the second dual tone multi-frequency signal, the network communication device is further configured to use the first audio signal according to the detected result. The part corresponding to the second dual tone multi-frequency signal is muted, and the first audio signal after the mute processing is output to the phone via the telephone adapter. 2. The video conferencing system of claim 1, wherein the network communication device comprises: a network module configured to establish communication with the network; and an audio module comprising: a keyboard event a transmitter, configured to convert the first command string into the first keyboard button according to the button conversion table; a dual tone multi-frequency signal forwarding module, configured to receive the first command word from the phone adapter String and forwarding the first command string to the keyboard event transmitter; an audio input module for receiving the first audio through the network 13 201240388 iwa ' 》 ^虎JU贞4 4th audio signal Whether the first frequency signal is included; a potential dual tone multi-frequency cancellation module for muting part of the second dual tone multi-frequency signal in the first audio signal; and an audio output module 'for The first audio signal after the mute processing is output to the phone via the telephone adapter. The invention further includes the subtraction conference system, wherein the first audio signal comprises a plurality of audio segments. The frequency input module has a first buffer and a second buffer, and the first buffer is configured to sequentially copy and temporarily store the first audio segments, and the first audio segments in the first buffer The length of the data is greater than a maximum detection length, and the first audio segments in the first buffer are combined in the second buffer to obtain a corresponding displacement value, and the audio input module cooperates with the displacement value to detect Detecting whether the first audio segments in the second buffer include the first-double s-frequency 3fL number, if the first audio segments include the second dual-tone multi-frequency signal, the dual-tone multi-frequency cancellation mode The set of the first audio segment in the second buffer is set to a threshold value. 4. The video conferencing system according to claim 3, wherein the audio input module is further _ the first Whether the first audio segments in the two buffers comprise the second dual tone multi-frequency _, and the number of times of over-reproduction is determined by the phone-period pulse code sampling frequency, the maximum side length, and a delay range constant Decided. 5. As stated in the third paragraph of the patent scope The conference system, wherein the length of the first audio segments set to a value of 0 is determined by a delay basic unit, a pulse code modulation sampling frequency per second of the telephone, and a length of the maximum detection 201240388. The video conferencing system of claim 2, wherein the live audio machine is further configured to generate a second audio signal, and the audio input module is further configured to detect whether the second audio signal includes a third dual tone. The second audio signal includes the third dual tone multi-frequency signal, and the two-tone frequency cancellation module is further configured to: according to the result detected by the audio input module, the second audio signal corresponds to the second audio signal. The part of the three-tone multi-frequency signal is subjected to static s processing, and is used to rotate the second audio signal after the mute processing to the network via the network module. The video conferencing system of claim 6, wherein the second audio signal comprises a plurality of second audio segments, the audio die and the eight buffers, the buffer and a second buffer, The first buffer is configured to sequentially copy and temporarily store the second audio segments, where the data length of the second audio segments in the first buffer is greater than the maximum debt measurement, and the first buffer The second audio segments are combined in the second buffer to obtain a displacement value, and the audio input module cooperates with the displacement value to form the second audio segments in the second buffer. ^ The second king doubles the voice, if the second audio includes the third dual tone multi-frequency signal 'the two-tone (four) cancellation module sets the second audio segments to a threshold value. In the video (4) system described in claim 7, the intermediate frequency input module further measures whether the second audio segment in the second buffer includes the third The dual-tone multi-frequency—the fork θ 夕 琥 , 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 、 Detecting length 15 201240388 1 W/j/4^A 9 The video conferencing system of claim 7, wherein the length of the second audio segment set to a value of 0 is determined by a delay basic unit One of the phones is determined by the pulse code modulation sampling frequency per second and the maximum detection length. A video conferencing method is applied to a video conferencing system, the video conferencing system includes a telephone, a telephone adapter, and a network communication device, the telephone adapter is connected to the telephone, and the network communication device is connected to The video adapter and the network, the video conference method includes: generating a first dual tone multi-frequency signal by using the phone; detecting the first dual tone multi-frequency signal by using the phone adapter to generate a first command The string VIII uses the network communication device to convert the first substring to a first-keyboard action to control an application according to a key conversion table; 接收 receiving a first via the network using the network communication device An audio message: and detecting whether the first audio signal includes a second dual tone multi-frequency network number including the second dual tone multi-frequency signal 'Using the second and second points of the section The part of the first audio signal corresponding to the second frequency signal is muted, and the mute processing s frequency signal is output to the telephone via the telephone adapter. In the I patent scope of the network, the video conferencing method of the 1G item, the road model:: for the display comprises a network module and an audio module, the network member transmitter network establishes communication, the audio module Including: a keyboard, a dual-tone multi-frequency communication, a gambling module, a bearing module and an audio output module, the video conferencing method 201240388 1 f ^ Ι-ΤΙ Γ\ includes: The multi-frequency signal forwarding module receives the first command string from the phone adapter and forwards the first command string to the keyboard event transmitter; - using the keyboard event transmitter to The first command string is converted into the first keyboard button action; the audio input module receives the first audio signal via the network, and detects whether the first audio signal includes the second dual tone multi-frequency signal; 4 * using the dual-tone multi-frequency cancellation module to mute the portion of the first audio signal corresponding to the second dual-tone multi-frequency signal; and using the "Hui frequency output module to mute the processing The first audio signal via the New Series connected to the various machines. 1 /2. The video conferencing method of claim 5, wherein the first audio signal comprises a plurality of first audio segments, and the audio input module has a first buffer and a second buffer. The visual method further includes: A. sequentially copying and temporarily storing the first audio segments by using the first buffer; and detecting a maximum length of the material length of the first audio segments in the first buffer Combining the first buffers of the first buffer with the second buffer and obtaining a corresponding one of the displacement values; and (4) the audio input module cooperates with the displacement to detect the first: inner: whether the first audio segment is Including the second dual-tone multi-frequency communication 17 201240388 ., 1 W / ΗΓΛ the first audio segment includes the second dual-tone multi-frequency signal, and the second buffer is used by the dual-tone multi-frequency cancellation module The first audio segments are set to a threshold value. 13. The video conferencing method of claim 12, further comprising: repeatedly detecting, by the audio input module, whether the first audio segments in the second buffer comprise the second dual tone multi-frequency The number of signals, and the number of repeated detections, is determined by one of the phone's pulse code modulation sampling frequency per second, the maximum detection length, and a delay range constant. 14. The video conferencing method of claim 12, wherein the length of time of the first audio segments set to a threshold is determined by a delay basic unit, one of the telephones, and a pulse code per second sampling frequency. And the maximum detection length is determined. 15. The video conferencing method of claim 11, further comprising: generating a second audio signal by using the audio input module or the phone; and detecting, by the audio input module, whether the second audio signal includes a third dual-tone multi-frequency signal; and the second second audio signal includes the third dual-tone multi-frequency signal, and the second-tone multi-frequency cancellation module is used to detect the second audio signal according to the result of the audio input module detection The part of the audio signal corresponding to the third dual-tone multi-frequency signal is statically processed, and the second sound-receiving signal after the mute processing is output to the network via the network module. The video conferencing method of claim 15, wherein the second audio signal comprises a plurality of second audio segments, and the audio input 201240388 input module has a first buffer and a second buffer. The video conferencing method further includes: sequentially copying and temporarily storing the second audio segments by using the first buffer; when the data length of the second audio segments in the first buffer is greater than a maximum detection length, Combining the second audio segments in the first buffer with the second buffer to obtain a corresponding one of the displacement values; and using the audio input module to detect the second buffer gains Whether the second audio segment includes the third dual tone multi-frequency signal; and the right first audio segment includes the third dual tone multi-frequency signal, and the second buffer is used by the dual tone multi-frequency cancellation module The second audio segments are set to a threshold value. 17. The video conferencing method according to claim 16, wherein the audio input module further detects whether the ϊγ Γ frequency segments in the second buffer comprise the third DTMF multi-frequency The signal, the domain is measured by the phone - the pulse is difficult to sample, the maximum detection length and a delay range constant. And the second application (4) _16 rhyme video conferencing method, Chi Meiben:: the length of the second audio segment of the value of 0 is determined by the - delay machine - per second pulse code modulation sampling frequency and the maximum
TW100109026A 2011-03-16 2011-03-16 System and method for video conference TWI448108B (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
TW100109026A TWI448108B (en) 2011-03-16 2011-03-16 System and method for video conference

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
TW100109026A TWI448108B (en) 2011-03-16 2011-03-16 System and method for video conference

Publications (2)

Publication Number Publication Date
TW201240388A true TW201240388A (en) 2012-10-01
TWI448108B TWI448108B (en) 2014-08-01

Family

ID=47599781

Family Applications (1)

Application Number Title Priority Date Filing Date
TW100109026A TWI448108B (en) 2011-03-16 2011-03-16 System and method for video conference

Country Status (1)

Country Link
TW (1) TWI448108B (en)

Family Cites Families (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5844979A (en) * 1995-02-16 1998-12-01 Global Technologies, Inc. Intelligent switching system for voice and data
US5920834A (en) * 1997-01-31 1999-07-06 Qualcomm Incorporated Echo canceller with talk state determination to control speech processor functional elements in a digital telephone system
TW563325B (en) * 2002-07-11 2003-11-21 Shinyea Comm Co Ltd Multi-party conferencing phone system and method thereof
US7546125B2 (en) * 2005-10-03 2009-06-09 Divitas Networks, Inc. Enhancing user experience during handoffs in wireless communication
CN201418113Y (en) * 2009-03-19 2010-03-03 何顺兰 Novel mini-type conference machine

Also Published As

Publication number Publication date
TWI448108B (en) 2014-08-01

Similar Documents

Publication Publication Date Title
US20060268836A1 (en) Method And Apparatus For Adapting A Phone For Use In Network Voice Operations
JP2006157120A (en) Network communication apparatus
KR20010084869A (en) Internet based telephone apparatus
CN105190752A (en) Audio transmission channel quality assessment
TW201021536A (en) Method, apparatus, and computer program product thereof for enabling an internet extension to ring a conventional extension
JPWO2004015972A1 (en) Voice communication system and method
TW201240388A (en) System and method for video conference
JP2007166393A (en) Ip telephone exchange apparatus
JP5696514B2 (en) Media communication apparatus, method and program, and media communication system
JP2011217005A (en) Intercom system, center device, and noise elimination method
JP2011239015A (en) Network apparatus and telephone system
JP4788553B2 (en) Network connection device
KR100396844B1 (en) System for internet phone and method thereof
JP5305533B2 (en) Communication test equipment
JP2007134862A (en) Ip communication terminal and ip communication method
JP5817898B2 (en) Media communication system, media communication apparatus, media communication method, and media communication program
JP2007329630A (en) Echo quality measuring apparatus, system, method, and program
TWI565291B (en) Telephone and audio controlling method thereof
KR0142799B1 (en) Volume control circuit in the phone
KR20140109111A (en) Audio output controlling method of communication device and apparatus thereof
US8204178B1 (en) Method to prevent TTY/TDD probing and unwanted TTY/TDD tone generation on voice gateways
CN201393255Y (en) Device for switching between IP phone and normal PSTN phone
JP2007165967A (en) Hybrid communication terminal and voice volume adjustment method of hybrid communication terminal
JP4375457B2 (en) Button telephone equipment
KR20010044591A (en) Sound card for internet phone