Real-time audio video regulation method

Real-time audio/video adjustment method

Technical field

The present invention relates to a method for transmitting and receiving audio/video (Audio/Video) data, and more particularly to a method for enabling a video conference system (Video Conference System) to transmit and receive during bandwidth reduction (Bandwidth) The method of audio/video data.

Background technique

For long-distance communications, the importance of video conferencing systems has gradually increased. With video conferencing systems, users can meet or talk through a system that includes two or more personal devices. The personal device consists of a viewing device (Viewing Device), a speaker (Speaker), a microphone (Microphone), and a video camera (Video Camera). The signal is transmitted through a communication device such as a telephone line (Telephone Line), a cable line (Cable), a satellite (Satellite), etc. At present, many video conferencing systems use the Internet to transmit audio and video data. The International Telecommunications Association (ITU) established the H.323 standard for network computers, instruments, and multimedia communication services to allow users to connect through the Internet and use different products that support the H.323 standard. The H.323 system can transmit real-time video, audio, or other data combined with elements. The H.323 standard defines how to format audio and video data, and how to package audio and video data for transmission over the network. Standard audio and video codecs encode or decode input/output analog data from audio and video signal sources into digital data for transmission on communication devices.

The main difficulty in transmitting audio, especially video signals, lies in the lack of sufficient bandwidth and latency. Bandwidth is a capacity measurement of the number of bits of data that can be transmitted per second. It is particularly important when an accurate amount of data must be provided within a specific period of time. Bandwidth problems can be caused by equipment, media, and basic equipment for network interconnection. In a video conference application with fixed audio and video data, for data transmission, whether the system has sufficient bandwidth is very important. For example, digital telephone quality speech (8-bit sampling, 8000 samples per second) requires approximately 64Kbps of data. For a Full Motion Video displayed by 1024 by 768 pixels, there are 3 colors for every 8 bits and 30 frames per second, which is equivalent to a capacity of 566,000. In a traditional system, when the transmission bandwidth is reduced during transmission, the system will reduce the video frame (Video Frame) to maintain a video system higher than the new low frequency bandwidth. This method is quite troublesome because the received video moves very fast. Therefore, in the transmission process, under the effective bandwidth reduction, it is urgent to transmit an audio/video data without affecting the video quality.

Summary of the invention

One of the objectives of the present invention is to provide a method for improving the conversion of video signals and audio signals during a reduced bandwidth. In one embodiment, it is used to convert audio and video signal data in a communication device. The conference system is the Internet. When the effective bandwidth is reduced during the data conversion period (when a conference is in progress), the video conference system will start to transmit and receive data with a quarter of the common media format QCIF instead of the common media format CIF to prevent The number of frames is reduced to maintain the continuity and quality of the video. The video conference device will receive the video signal data in QCIF format and convert it into CIF format data for observation. In another embodiment of the present invention, if the bandwidth decreases, the audio format can be changed according to the amount of bandwidth. In another embodiment of the present invention, the user can choose his or her preferences, and based on the method of the present invention, the audio quality can be reduced while preserving the video quality, or conversely, the video can be transmitted in QCIF format to preserve the audio quality .

Description of the drawings

Figure 1 shows a typical video conferencing device.

Fig. 2 is a flowchart according to an embodiment of the present invention.

Fig. 3 is a flowchart according to another embodiment of the present invention.

Description of reference signs: 10: Audio-video signal input and output 20: Processor 30: Transmitter 40: Communication device 100-140: Steps of the first embodiment 200-270: Steps of the second embodiment 2 is a typical device of the present invention. A processor 20 is connected to an audio-video input and output 10, a typical video output is a liquid crystal display screen (LCD Screen), and the audio output can be a speaker. The video input can be a camera, and the audio input can be a microphone. The processor 20 uses a transmitter (Transmitter) 30 to convert the input signal into a signal that can be propagated on a communication device. The communication device 40 includes standard PSTN lines, radio communication (Radio Communication), cables, and the like. A receiver (Receiver) 30 transmits the received signal to a processor 20 that converts the digital signal into a signal for A/V output 10. The personal computer can obtain the functions of the video conferencing system through appropriate hardware (Hardware). This system can also be used as a Multipoint System that incorporates a Multipoint Control Unit (MCU) to allow 3 or more H.323 endpoints to connect and participate in a conference.

The ITU standard H.261 defines the video codec for audio-visual services in P×46Kbps HTML. H.261 defines two common media format (CIF) image formats, and a quarter of the common media format (QCIF). CIF supports an image resolution of 352×288 pixels per frame, while QCIF supports an image resolution of 176×144 pixels per frame (a quarter of CIF). QCIF/CIF video can be encoded using the YcbCr color coding scheme (Y=luma, Cb/Cr=Chrominance Color Encoding Scheme). The advantage of this method is that the human eyes are less sensitive to color changes than to intensity, so that the color component (Cb/Cr) can be encoded at half the brightness bandwidth and the color quality can be maintained.

The present invention provides a method for transmitting video and audio data during the effective period of low frequency bandwidth. This system can monitor the video bandwidth at the connection point, and can also monitor the video bandwidth when using Instant Transmission Contract (RTP) and Real-time Transmission Control Contract (RTCP) to determine the state of network traffic. For example, if the complete RTP wave packet and the complete RTCP wave packet are not received, it can be deduced that the bandwidth is inappropriate. Or, if the RTCP wave packet is received without frame loss, the video or audio codec can be adjusted to reduce the required bandwidth. If the RTP wave packet is received without frame loss, but the RCTP wave packet is lost, because there is no loss of video or audio frames, it can still be regarded as an acceptable bandwidth. Once the RTP wave packet and RTCP wave packet are received, and there is no frame loss, it is known that the bandwidth is appropriate, and it can be decided to adjust the video and audio codec to obtain better quality, and then the bandwidth can be determined to be effective. To take appropriate action on the system.

In an embodiment of the present invention, if the effective bandwidth is reduced during the conference, the video conference system (including two or more personal units) will transmit and receive video data in QCIF format instead of CIF format . Each video conference unit can independently start transmitting QCIF when the bandwidth is reduced to a certain value. Alternatively, during the transmission process of the video conference unit, it may be negotiated or required to start transmitting and receiving QCIF video data at the same time.

In order to transmit the video data in the QCIF format, the processor can directly convert the input signal from the A/V input 10 into a QCIF format. In another embodiment, the processor may compress data in CIF format to conform to QCIF. Then a unit can receive data in QCIF format and convert the image into CIF format for display. Figure 2 is a flow chart of the method. As usual, an audio-video system starts transmission in the CIF format in step 100. If during transmission, the bandwidth drops below the first predetermined level, the system starts transmission in the QCIF format in step 110. Similarly, the receiving station will start to receive the QCIF signal and convert it to a CIF format again (step 120 ). If the bandwidth is increased later, the system can start transmitting video data in CIF format again (step 140).

In another embodiment of the present invention, if bandwidth reduction occurs, the audio format can be changed. As usual, an audio-video signal system can use a codec, such as G.711 that transmits 48, 56 or 64 Kbps. However, if the bandwidth is reduced during transmission, the system can use a different low-bandwidth codec, such as G.728 (16Kbps) or G.723 (5.3 or 6.3Kbps). As shown in the flow chart of Fig. 3, the system uses the G.711 codec in step 200 to start transmitting audio data. If the bandwidth drops below a first predetermined level during transmission, the system may start G.728 transmission in step 220. If the bandwidth increases, the system can start transmitting G.711 again. However, if the bandwidth drops below a second predetermined level (step 240), the system will start transmitting G.273 audio in step 250.

In another embodiment of the present invention, according to the user's preference, the audio format can be changed to reduce the low frequency bandwidth and maintain the video quality in the display method. Conversely, the video data bandwidth can be reduced by switching from the CIF format to the QCIF format, so that the audio quality can be maintained based on this method. Also, the audio and video data bandwidth can be reduced to keep both signals at an appropriate level. For example, if it is detected that the bandwidth is reduced, the audio format can be changed to G.728. If there is still a lack of effective bandwidth, the video data format can be changed to the QCIF format to prevent the audio format from having to be changed to G.273. Using this method, a compromise can be achieved between audio and video data to keep the two at an appropriate level.

Although the present invention has been described above with examples, it is not intended to limit the present invention. Anyone familiar with this technology can make various changes and modifications without departing from the spirit and scope of the present invention. The scope of protection of the invention shall be subject to the claims.