Abstract:
A webcasting system and the audio data regulating methods to be used in the webcasting system are presented. The webcasting system includes a host and an audio playing apparatus. The host, which is loaded with an operating system and drivers, determines the audio data output according to an expected data received by the operating system. The drivers provide the expected data according to the audio data received and transform the audio data for network transmission. The audio playing apparatus receives the network data and processes the network data for audio playing.

Description:
This application is a divisional application of co-pending U.S. application Ser. No. 11/656,409, filed Jan. 23, 2007. This application claims the benefit of Taiwan application Serial No. 95102518, filed Jan. 23, 2006, the subject matter of which is incorporated herein by reference. 
    
    
     BACKGROUND OF THE INVENTION 
     1. Field of the Invention 
     The invention relates in general to an audio processing method, and more particularly to an audio regulating method used in network audio playing. 
     2. Description of the Related Art 
     When audio is being played in a host having a sound card, a mixing program, such as a core-mixing program of an operating system (KMixer: kernel mixer of Microsoft Windows) is utilized to control the transmission quantity of audio data. After that, a hardware channel interface driver, such as WavePci or WaveCyclic of the miniport driver in Microsoft Windows, receives the audio data transmitted from the mixing program and then outputs the audio data to the sound card for playing the audio. The mixing program provides the quantity of data transmission precisely when the hardware channel interface driver provides the playing position of the audio data. That is, the mixing program can correctly control the transmission quantity per second of the audio data with the data playing position given by the hardware channel interface driver. 
     However, if audio is broadcasted by way of webcasting, a virtual sound card is utilized to receive the audio data since the host does not have a physical sound card, and the audio data is then transmitted to an audio playing device through the Internet. The hardware channel interface driver can calculate the data quantity of audio data required in a certain period of time according to the time accumulated during this certain period of time. Thus, the mixing program can control the data quantity of audio data transmitted to the hardware channel interface driver for further handling. For example, in a Microsoft operating system, the quantity of data transmission is obtained by using Getposition( ) of the IMiniportWaveCyclicStream or IMiniportWaveCyclicStream interface. In the example of a virtual audio driver, the implementation of Getposition( ) is to get the time difference between the current time and the previous time of calling Getposition( ), in order to get the quantity of audio data transmission. Then, the approximate playing position is calculated according to the format and the transmission rate of the audio data such that the mixing program can refer to and control the quantity of audio data transmission. However, when being used in the network for real time audio playing, the approximate playing position and the error accumulated during multiple calculations will cause unpleasant crackling sounds during real time audio playing. 
     SUMMARY OF THE INVENTION 
     The invention is directed to a method of regulating audio data to prevent the crackling sounds caused by the unprecise supply-demand flow of the audio data when the audio is being played. 
     According to a first aspect of the present invention, an audio data transmitting apparatus is provided. The apparatus includes a storage unit, an output unit and a regulating unit. The regulating unit includes a first layer program and a second layer program. The first layer program regulates an output quantity of audio data according to expected data. The second layer program calculates a first data quantity of the output quantity of the audio data between a real time instant and a base time instant, calculates a difference between the first data quantity and a second data quantity, transmits the difference back to the first layer program, and transforms the audio data transmitted from the first layer program into virtual audio data. The difference is the expected data, and the second data quantity is the output quantity of the audio data between the previous real time instant and the base time instant. The storage unit stores the virtual audio data. The output unit transforms the virtual audio data into transmissible data with a transmissible format. 
     According to a second aspect of the present invention, an audio regulating method is provided. The method includes the following steps. First, the method receives audio data. Next, the method regulates an output quantity of the audio data according to expected data. Then, the method calculates a first data quantity of the output quantity of the audio data between a real time instant and a base time instant, calculates a sum of the first data quantity and a second data quantity, and transmits the sum back to a first layer program. An integer part of the sum is the expected data. The base time instant is a previous real time instant. The second data quantity is a fractional part of a previous sum. 
     The invention will become apparent from the following detailed description of the preferred but non-limiting embodiments. The following description is made with reference to the accompanying drawings. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         FIG. 1  shows the architecture of a network audio playing system according to an embodiment of the invention. 
         FIG. 2  is a schematic illustration showing time instants for audio data outputting. 
         FIG. 3  is a flow chart showing a method of regulating an audio data quantity according to one embodiment of the invention. 
         FIG. 4  is a flow chart showing a method of regulating an audio data quantity according to another embodiment of the invention. 
     
    
    
     DETAILED DESCRIPTION OF THE INVENTION 
       FIG. 1  shows a webcasting system  100  according to an embodiment of the invention. Referring to  FIG. 1 , the webcasting system  100  includes an audio data transmitting apparatus  110  and a playing device  120 . The audio data transmitting apparatus  110  transforms audio data S 1  into network data S 4  and outputs the network data S 4  to an audio playing apparatus  120 , which plays the audio according to the network data S 4 . 
     The audio data transmitting apparatus  110  is loaded with a first layer program  111  and a second layer program  112 . The second layer program  112  receives the audio data S 1  transmitted from the first layer program  111 , and then transmits the expected data S 2  of the audio data S 1  back to the first layer program  111 . The first layer program  111  controls the data quantity of the audio data S 1  to be outputted to the second layer program  112  according to the expected data S 2 . The audio data transmitting apparatus  110  runs the second layer, program  112  to transform the outputted audio data S 1  into the network data S 4 . 
     The first layer program  111  and the second layer program  112  are respectively a core-mixing program and a miniport driver in this embodiment, which are both found in the Microsoft Windows operation system. A sub-program Getposition( ) of the second layer program  112  gets the expected data S 2  and then transmits the data S 2  back to the first layer program  111  such that the first layer program  111  can control the output data quantity of the audio data S 1 . 
     After receiving the audio data S 1 , the second layer program  112  transforms the audio data S 1  into virtual audio data  53  and transmits the virtual audio data S 3  to a virtual sound card  113 . A network adapter  114  transforms the virtual audio data of the virtual sound card  113  into network data S 4  and outputs through the network. The virtual sound card  113  is a storage unit in the audio data transmitting apparatus  110 . The network adapter  114  serves as an output unit for transforming the virtual audio data S 3  into the network data S 4  with a transmissible format and then outputting the network data S 4  to the audio playing apparatus  120  through wired or wireless network. 
     The audio playing apparatus  120  includes a network adapter  121 , a sound card  122  and an amplifying speaker  123 . The network adapter  121  serves as a receiving unit for receiving the network data S 4  through the network and disassembling the packets of the network data S 4  into a virtual audio signal S 5 . Then, the sound card  122  generates playable audio data S 6  for the audio playing unit according to the virtual audio signal S 5 , and the amplifying speaker  123  plays the audio data S 6 . The sound card  122  of the playing device  120  is a physical sound card. 
     In order to enable the second layer program  112  to provide the desired transmission data quantity of the audio data S 1  for the first layer program  111  during the actual playing procedure to prevent the audio playing apparatus  120  from generating the crackling sounds due to the inconsistency between the audio data quantity and the audio position when the audio is being played at the end of the audio data transmitting apparatus  110 , several methods for obtaining the transmission data quantity of the audio data are provided to solve this problem. 
       FIG. 2  is a schematic illustration showing time instants for audio data outputting. As shown in  FIG. 2 , when the second layer program  112  is calculating the expected data S 2  of the audio data S 1 , a selected base time instant BT is compared with a real time instant to get the expected data S 2  in order to prevent the error in the data quantity. As shown in  FIG. 2 , the current play time instant is the real time instant T 1 , and the previous real time instant is T 0 . The instants T 1  and T 0  are respectively compared with the base time instant BT, and then the transmission data quantity of the audio data S 1  (i.e., the expected data S 2 ) between the real time instant T 1  and the previous real time instant T 0  can be obtained and transmitted back to the first layer program  111 . The previous real time instant T 0  is the previous time instant of calculating the transmission data quantity of the audio data S 1 . 
     Please refer to  FIGS. 2 and 3  simultaneously.  FIG. 3  is a flow chart showing a method of regulating an audio data quantity according to one embodiment of the invention. 
     First, step  31  obtains the base time instant BT. Next, as shown in step  32 , the first data quantity D 1  is calculated according to the time difference TD 1  between the real time instant T 1  and the base time instant BT, wherein the first data quantity D 1  is the transmission data quantity of the audio data S 1  between the real time instant T 1  and the base time instant BT. 
     As shown in step  33 , the zeroth data quantity D 0  is calculated according to the time difference TD 0  between the previous real time instant T 0  and the base time instant BT, and then the difference (i.e., the second data quantity D 2 ) between the audio data output quantity (i.e., the first data quantity D 1 ) and the previous audio data output quantity (i.e., the zeroth data quantity D 0 ) is obtained. 
     As shown in step  34 , the expected data S 2  is generated according to the second data quantity D 2  such that the first layer program  111  can regulate the transmission data quantity of the audio data S 1  according to the expected data S 2 . 
     As shown in  FIG. 2 , the zeroth data quantity D 0 , the first data quantity D 1  and the second data quantity D 2  are respectively obtained by multiplying the time difference TD 0 , the time difference TD 1  and the time difference TD 2  by a bitrate in steps  32  and  33 . 
     In this embodiment, the base time instant may be reset after a specific period of time, in order to prevent the prolonged audio playing procedure from causing the overflow problem, as shown in step  36 . For example, a new base time instant BT′ is set to replace the base time instant BT after a specific period of time, and the expected data S 2  is calculated according to the base time instant BT′. 
     In this embodiment, the expected data S 2  is calculated according to the calculation made between the base time instant BT and the real time instant T 1 . So, the crackling sounds existing in prior art due to approximation of play position resulting from multiple times of error accumulation is avoided. 
     In another embodiment, the expected data S 2  is directly obtained according to the relative time difference between the current time instant and the previous time instant of calculating the transmission data quantity of the audio data S 1 . The value of the expected data S 2  should be an integer in terms of a data unit, such as 1 byte. Thus, this embodiment accumulates and records the remainders of the transmission data quantity that are smaller than one data unit during each calculation, in order to prevent the error accumulation caused by multiple times of skipping the fractional part. 
       FIG. 4  is a flow chart showing a method of regulating an audio data quantity according to another embodiment of the invention. First, as shown in step  41 , the newest audio data output quantity D is obtained from the time instant of obtaining the previous audio data output quantity. As shown in  FIG. 2 , the audio data output quantity D is the quantity of audio data output from the previous real time instant T 0  to the real time instant T 1 . Next, in step  42 , the audio data output quantity D obtained in step  41  and the originally accumulated output quantity DT are summated. The accumulated output quantity DT is the fractional part of the audio data output quantity from the base time instant BT to the previous real time instant T 0  in  FIG. 3 . Thereafter, in step  43 , the expected data S 2  is generated according to the integer part [DT] of the sum to regulate the output of the audio data S 1 . In step  44 , the accumulated output quantity DT is updated to be the fractional part DT-[DT] of the sum repeatedly until the end of the audio is reached. 
     If the originally newest audio data output quantity D is 2.13244 bytes and the originally accumulated output quantity DT is 0.6 bytes, the sum is 2.73244 bytes. In step  43 , the integer part [DT] (=2 bytes) of the sum is taken as the expected data S 2 . In step  44 , the accumulated output quantity DT is recorded as the fractional part DT-[DT] (=0.73244 bytes) of the sum to serve as the reference of accumulation when the expected data S 2  is obtained at a next time. 
     If the originally newest audio data output quantity D is 2.13244 bytes and the originally accumulated output quantity DT is 0.9 bytes, the sum is 3.03244 bytes. In step  43 , the integer part [DT] (=3 bytes) of the sum is taken as the expected data S 2 . In step  44 , the accumulated output quantity DT is recorded as the fractional part DT-[DT] (=0.03244 bytes) of the sum to serve as the reference of accumulation when the expected data S 2  is obtained at the next time. 
     According to the webcasting system and the audio regulating method according to the embodiments of the invention, the audio can be played through the network, and it is also possible to control the network audio playing device to play the audio by way of wireless networks. Since the end of the audio data transmitting apparatus can provide the displacement of the audio data precisely, the errors of the data and the play position will not accumulate, and the crackling sounds caused by insufficient data quantity are thus avoided. 
     While the invention has been described by way of examples and in terms of preferred embodiments, it is to be understood that the invention is not limited thereto. On the contrary, it is intended to cover various modifications and similar arrangements and procedures, and the scope of the appended claims therefore should be accorded the broadest interpretation so as to encompass all such modifications and similar arrangements and procedures.