74results about How to "Guaranteed playback quality" patented technology

The invention provides a network streaming media transmission disconnection re-entering technique. The technique includes steps as follows: (1) detecting network disconnection; (2) detecting network connection state; (3) receiving date continuously. Client calculates program NPT corresponding to network disconnection time according with time relationship in streaming media system, then, client uses the time as one parameter of RTSP agreement PLAY method after NPT requiring program disconnection to streaming media server, and realizes streaming media loading or playing continuously. The PRSP agreement OPTIONS method and PLAY method provided by the invention can make client receive date from server continuously when network connection is returned to normal after client network disconnection, and ensure loading integrity and playing quality of streaming media date in client.

The invention discloses a multipoint control unit (MCU) cascade system which comprises a plurality of MCU under the management of a service management center. The cascade manner among each MCU in the system is reticulate cascade. The invention discloses a communication method for the multipoint control unit (MCU) cascade system, the multipoint control unit (MCU) and a service management center simultaneously. In the embodiment of the invention, as a connection networking manner among each MCU in the cascade system is changed, minimum transmission paths though the MCU can be selected according to the connection situation of the MCU, thus being capable of effectively avoiding transmission delay and the reduction of video and audio quality caused by multiple transmitting of the MCU, reducing transmitting time of media streams on the whole and ensuring the broadcasting quality of the media streams.

The present invention provides a method and apparatus for configuring software resource for playing network programs. A request for downloading network programs transmitted by a user terminal to a network server includes the configuration information of software resources at the user terminal. After comparing and balancing the software resources required for playing network programs and the software resources at the user terminal, the network server transmits to the user terminal the software resource that is required for playing programs but are absent at the user terminal. The user terminal combines the received software resource with the local software resources to play the downloaded network programs. The user terminal can sufficiently configure the software resources required for playing network programs only by attachedly transmit the configuration information of software resources that the user terminal has when transmitting a request for downloading network programs.

The embodiment of the invention discloses a video polyphonic ringtone playing method which comprises the following steps: under the condition that a calling request of a calling terminal is received,determining a video polyphonic ringtone file of the calling terminal; Obtaining video parameters of the video polyphonic ringtone file; Sending the video parameters to the calling terminal; Wherein the video parameters are used for indicating the calling terminal to prepare for playing the video polyphonic ringtone file. Therefore, the terminal can prepare for playing the video polyphonic ringtonefile according to the video parameters received in advance, so that the playing success rate and the playing quality of the video polyphonic ringtone file are ensured. The embodiment of the inventionalso discloses a video polyphonic ringtone playing device and a video polyphonic ringtone playing system.

The invention is applicable to the technical field of web televisions and provides a method for intelligently playing a web television. The method comprises the following steps of: 10, starting the web television; 20, judging whether an intelligent play function is opened, if so, turning to a step 30; 30, detecting the network speed; 40, dividing a network speed grade according to the network speed; and 50, taking an audio file as a preferentially played file according to the network speed grade, and selecting a playing mode matched with the network speed. Due to the method for intelligently playing the web television, the playing of the audio file is preferentially guaranteed, so that a user can hear smooth sound at any network speed, and the playing quality is guaranteed.

The invention relates to a method of real-time flow media service distribution resource, wherein it comprises: the group data service node catches the flow media service report sent by user terminal; analyzing said report to judge the start of real-time flow media service, to initialize the refresh of data pack protocol PDP context, and perform resource for said real-time flow media service; judging the end of said service, to initialize the refresh of PDP context, and recycle the preformed resource. The invention also provides a relative device, which comprises: a detecting / analyzing module for catching and analyzing the service report sent from user terminal, a match judging module for matching the flow media service and the speed, and a resource manage module for performing relative resource for the real-time flow media service, based on the speed. The invention can confirm the service quality of flow media service and improve the network resource utilization.

The invention discloses a method and system for quickly starting playing of network multimedia files. The method comprises the steps that multimedia file metadata and key media content of server sides are pre-stored in client sides, the pre-stored multimedia file metadata are indexed, the multimedia files which are requested to play by a user are obtained, and the content steps or modules which are requested to play by a user are obtained, and the pre-stored multimedia file metadata and key media content between the server sides in the client sides are updated and synchronized. The system comprises a module used for pre-storing the multimedia file metadata and the key media content of the server sides to the client sides, a module for indexing the pre-stored multimedia file metadata and obtaining the multimedia files which are requested to play by the user, and a module for updating and synchronizing the pre-stored multimedia file metadata and the key media content between the client sides in the server side. Through the method and the system, the user can enable the network multimedia files to be played and started more quickly, play quality is guaranteed, low re-cache rate is maintained, and using experience of the user is further improved.

The invention discloses a video resource transmission parameter distribution method and device. The method comprises the following steps of during a video live broadcast process, obtaining a to-be-transmitted video resource in a video live broadcast stream; detecting a target video scene corresponding to the video resource, the target video scene being a scene to which content played in the videoresource belongs; distributing a target transmission parameter which corresponds to the target video scene and satisfies a target code limiting threshold value to the video resource, wherein the target coding limiting threshold value is the limiting range of the live broadcast channel for the transmission parameters of the video live broadcast stream, the live broadcast channel is used for transmitting the video live broadcast stream, and the target transmission parameters are the transmission parameters used when the video resources are coded. According to the invention, the technical problemof worse flexibility of transmission parameter distribution of video resources in the prior art is solved.

The invention relates to a self-adaptive P2P (Peer-to-Peer) steam media data piece selection method and a node. The method is carried out based on collaborative nodes in a P2P network; the collaborative nodes contain data pieces requested by the current node for scheduling. The method comprises steps of: estimating the priority of each data piece to be requested of the first node; obtaining the total priority of each data piece to be requested by the first node according to the proportional relation among the data pieces based on the priority, and sequencing all data pieces according to the total priority; selecting a plurality of previous continuous data pieces to be requested in the current scheduling period by the first node according to the sequence of the total priority, and requesting; and then providing the data piece selected as the server requested to adjacent nodes or media, wherein the first node is the node for sending a request on the steam media data piece to the collaborative adjacent nodes. The priority of the data piece includes the sequence scheduling priority, the rare scheduling priority, and the random scheduling property of the data pieces, which are calculated to obtain the final priority of the data pieces.

A video / audio display system includes a buffering module, a playing module and a controlling module. The buffering module temporarily stores video / audio data, which occupy a storage capacity of the buffering module. The playing module is electrically connected with the buffering module to play the video / audio data at a playing speed. The controlling module is electrically connected with the buffering module and the play module. A video / audio display method is also disclosed.

The invention discloses a network-coding-based fair service method for the streaming media of a wireless MESH network. In the wireless MESH network, the broadcasting characteristics of a wireless channel and the advantages of network coding are combined, the value of coded data is fully found, and a fair service strategy is adopted, so that each target node can well enjoy the fair service of the streaming media. By the method, the rate of data forwarding from an intermediate node to the target node can be balanced, the throughput of the wireless MESH network can be improved, and the play quality of the streaming media can be improved.

The invention method resolved problems in prior art technology of: wasting manpower resource, unsatable playing quality and high cost. The collecting end with collecting software and the transmitting end and connected with control computer with controlling software by communication method. The digital video informatino collected by collecting end are edited by editors at control computer end, and the of digital video informations on transmitting end is controlled by editors at control computer end, thus with the advantages of: saving manpower, stable broadcasting quality and low cost.

The embodiment of the invention provides a data processing method and device and a smart TV system for solving the problem of resource waste resulting from frequent hardware update caused by update. The method comprises the following steps: receiving transport stream pushed by service providing equipment allocated by a network side by a client; analyzing the transport stream to a corresponding user interface, and displaying the user interface; after a multimedia playing request is triggered based on the user interface, receiving multimedia playing information fed back by the service providing equipment according to the multimedia playing request; and sending a first instruction to the service providing equipment to cause the service providing equipment to pause to continue to push the transport stream, and playing corresponding multimedia data according to the multimedia playing information. Update and other contents are executed on the side of the service providing equipment, analysis and display are carried out on the side of the client to prevent the resource waste resulting from the frequent hardware update, and the service providing equipment pushes multimedia resources without employing the transport stream, thereby guaranteeing the playing quality of the multimedia resources while saving the bandwidth resources.

The embodiment of the invention discloses a sound signal processing method. The sound signal processing method comprises the steps of obtaining a first sound signal acquired by a microphone of a mobile terminal; calculating the short-time zero-crossing rate of the first sound signal, wherein the short-time zero-crossing rate comprises the zero-crossing rate of sounds within 5ms-15ms; removing a low-frequency part and a high-frequency part of the first sound signal so as to acquire a second sound signal; acquiring energy data of the second sound signal; removing a non-speech part of the second sound signal according to the short-time zero-crossing rate and the energy data so as to acquire a third sound signal; acquiring a whistle signal of the third sound signal; removing a single-frequency signal of the whistle signal in the third sound signal so as to acquire a fourth sound signal; and amplifying the fourth sound signal. The invention further discloses a corresponding mobile terminal. The sound signal processing method disclosed by the embodiment of the invention realizes combined human sound recognition, ensures the play quality of a human sound while suppressing or removing whistle sounds, and acquires better user experience.

The invention provides a method and system for achieving audio and video synchronous speed variation. The method comprises the steps that audio frequency with speed to be varied in a first audio cache region is read, and audio parameters of the audio frequency with the speed to be varied are extracted; a speed variation coefficient is obtained, and the audio quality of the audio frequency of which the speed to be varied is varied according to the speed variation coefficient is evaluated; when the audio quality does not reach a reserved requirement standard, exception processing is conducted on the audio frequency with the speed to be varied; when the audio quality reaches the reserved requirement standard, processing with speed variation and without tone variation is conducted on the audio frequency with the speed to be varied according to the speed variation coefficient, and then synchronization is conducted on corresponding video which is subjected to speed variation according to the speed variation coefficient. By means of the method and system for achieving the audio and video synchronous speed variation, audio and video synchronous speed variation can be achieved when an audio and video file is played in a speed variation mode, tone and timbre of the audio frequency can be kept invariant, and the playing quality of the audio and video file which is subjected to speed variation is guaranteed.

The embodiment of the invention discloses a video playing method. The video playing method comprises the following steps: detecting a network type of a network where a terminal is located at present; judging whether the network type of the network where the terminal is located at present is a preset network type; when it is determined that the network type of the network where the terminal is located at present is the preset network type, downloading a video file according to a preset video playing list; and when a video playing request is received, playing the video file selected by the video playing request. The invention further discloses a video playing apparatus and a terminal. By adopting the video playing method and apparatus, the terminal can download the video file in advance and then play the same to effectively avoid the situation that the terminal needs to pay when viewing an online video in a paid network environment, thereby reducing the use cost of users, guaranteeing the playing quality and improving the user experience.

The invention discloses a video encoding and decoding method and device and a storage medium. The method comprises the following steps: determining a to-be-decoded video frame; decoding the video frame from the decoded video frame; obtaining a reference decoding parameter of a reference frame associated with the to-be-decoded video frame, wherein the reference decoding parameter comprises at leasttwo of the following parameters:, the number of blocks decoded in a skip mode in the reference frame and the total number of blocks in the reference frame, the first number of blocks decoded in an intra-frame and the second number of blocks decoded in an inter-frame in the reference frame, an energy parameter of a motion vector in the reference frame, the third number of blocks of a first size inthe reference frame and the fourth number of blocks of a second size in the reference frame; and the first size is smaller than the second size; determining a target resolution for decoding the to-be-decoded video frame according to the reference decoding parameter; and decoding the video frame to be decoded by adopting the target resolution. According to the invention, the technical problem of low video encoding and decoding efficiency caused by complex processing operation provided by related technologies is solved.

The invention provides a method and a device for playing a digital audio which are applied in a digital audio broadcasting receiving terminal with local clocks. The method comprises the steps of counting the number of the local clocks in audio data frames, each audio data frame comprises a plurality of audio data; obtaining the number of the local clocks in the audio data according to the number of the local clocks in the audio data frames; counting the local clocks, generating a playing signal when counting the number of the local clocks in the audio data and outputting the audio data according to the playing signal for playing the audio. The invention enables the digital audio broadcasting receiving terminal to accurately obtain the audio playing frequency, leads the jitter of the audioplaying frequency to be reduced to the minimum and further ensures the playing quality of digital audio broadcasting.

The invention relates to an archive system VFS design algorithm of real-time media data. The disk space is automatically divided and formatted by a file system, and independent sub regions and disk information are maintained by software. The access of the data is processed in a way according to large data blocks, and in addition, files are distributed and stored in continuous disk positions. The user service quality can be ensured through calculating the band width capable of being supported by the disk. Vacant nodes are managed by a vacant node chain table of the disk structure for the space distribution of the files. The file writing space is pre-distributed, after the pre-distribution, the dynamic regulation is carried out, in addition, the most proper space is found in the vacant node chain table of the disk structure to be distributed.

The invention applies to the technical field of communication and provides a correction method and device for a frame loss rate and a terminal. Through utilization of the correction device for the frame loss rate, after a computing mode of the frame loss rate is determined according to an acceleration mode variable, when it is determined that a current congestion code frame is not a frame I or a first frame P following the frame I, the frame loss rate is computed according to the determined computing mode. A current congestion code is coded according to the frame loss rates obtained through computing based on different computing modes, so the coding efficiency is improved, moreover, the fluency degree of a video is ensured, the playing quality of the video is ensured, the balance between the fluency degree and the playing quality is balanced, and the user experience is relatively good.

The invention relates to the technical field of communication, in particular to a method for transmitting multimedia signals in a tramcar. The method includes the steps that firstly, AV signals generated by a ground control center are received by an AV signal receiver, an AV circuit is adopted between the AV signal receiver and a control panel interface for transmission, analog-digital conversion is conducted on the AV signals through a control panel, the AV signals are converted into digital signals which are transmitted through a network, when the digital signals are transmitted through the network, firstly, corresponding audio compression and video compression are conducted on the AV digital signals, and an AV synchronization algorithm is adopted in compressed AV data. The method solves the problems that AV signal remote transmission lines are difficult to arrange and signals interfere with one another, and the method guarantees the AV playing quality.

The loudspeaker module comprises an elastic piece, a vibrating diaphragm assembly, a driving assembly and a basin stand, the elastic piece further comprises an outer fixing part, a central elastic piece located on the inner side of the outer fixing part and a connecting arm connected with the outer fixing part and the central elastic piece, the outer fixing part is connected with the basin stand, the central elastic piece is connected with the driving assembly, and the central elastic piece is connected with the driving assembly. The central elastic piece is further connected with a transmission part, the vibrating diaphragm assembly is connected with the transmission part, the driving assembly is controlled to drive the central elastic piece and the connecting arm to vibrate, the vibrating diaphragm assembly is driven by the transmission part to synchronously vibrate and produce sound, the connecting arm is further connected with a balance weight, and the compliance of the connecting arm and / or the weight of the balance weight are / is used for adjusting the output characteristic of vibration sound production. According to the loudspeaker module provided by the invention, the output characteristic is adjusted through the elastic sheet and the counterweight, the loudspeaker module is easy to implement in a thin space, and the low-frequency response performance is improved while the medium-high frequency playback quality is ensured, so that the full-band sound playback quality of the loudspeaker is improved.

The invention discloses a set-top box and a set-top box safety protection method. The problem of heat accumulation caused by current flowing on a cable television network and power network ground wire is avoided. The set-top box comprises a main processing circuit, a cable television signal receiving terminal, a gating circuit, a through circuit and an isolation circuit. When the set-top box detects a startup signal and judges that current on the cable television signal ground wire exceeds a threshold value, the gating circuit is controlled to be communicated with the output of a second output end, and the isolation circuit isolates the current of the ground wire. If the current does not exceed the threshold value, the gating circuit is controlled to be communicated with the output of a first output end, so that the through circuit is communicated with the cable television signal ground wire. After a shutdown signal is detected, the gating circuit is controlled to be connected with a third output end, and the channel of the cable television signal ground wire is cut off. In the aspects of startup and shutdown, after the current of the ground wire between networks is large due to poor or unauthorized wire building, the channel of the ground wire current is cut off, and safety risks caused by long time current accumulation are eliminated.

The embodiment of the invention provides a video data downloading method and device, electronic equipment and a storage medium, and the method comprises the steps: downloading first sub-video data before a first downloading progress in target video data through a CDN when a first video-on-demand request for the target video data is obtained; downloading second sub-video data between the first downloading progress and the second downloading progress in the target video data through the first P2P network; obtaining a cache time length; according to the cache duration, a first target network is determined, third sub-video data located after the second downloading progress in the target video data is downloaded through the first target network, and the first target network comprises at least one of the following items: a first P2P network, a second P2P network and a CDN. According to the invention, network switching can be realized in the downloading process, so that the traffic cost can be controlled while the video playing quality is ensured.

The invention discloses a live broadcast data processing method and data processing equipment based on an HLS protocol and a computer readable storage medium. The method comprises the steps: acquiring an M3U8 list, and determining the buffer time of a TS fragment to be downloaded corresponding to the M3U8 list; when the buffer time does not reach the safety time, acquiring a point-to-point sharing list, and determining the probability of downloading the TS fragment to be downloaded through a CDN according to the point-to-point sharing list; when the probability is smaller than a preset threshold value, downloading the TS fragment to be downloaded in a PCDN back-to-source mode; and storing the TS fragment to be downloaded, so as to enable a playing terminal to acquire the downloaded TS fragment through the processing equipment. The invention aims to achieve the effect of further improving the sharing rate on the basis of ensuring the playing quality.

The present invention relates to the technical field of video conferences and particularly to a video conference system. A microprocessor is electrically connected with a starter, a storage device, and an information collecting module; the information collecting module is electrically connected with a video collector and a microphone module; the video collector is electrically connected with a camera, and the camera and the microphone module are both arranged in a conference room; the microprocessor is further electrically connected with a noise cancellation module, a data conversion module and an information receiving and transmitting module; the microprocessor is further electrically connected with an audio processing module and a video processing module, and the audio processing moduleis electrically connected with a loudspeaker; and the video processing module is electrically connected with a display screen. The video conference system in the invention realizes multi-party conferences in different regions, can collect live scenes and audios of different conference sites in real time, and can effectively remove noise in the live scenes and audios, thereby realizing conference sharing in different regions.

The invention discloses a video playing method and device and electronic equipment, and belongs to the technical field of communication. The method comprises the following steps: acquiring video frame data of a target video and displacement data of a target object in a video frame; when the displacement data indicates that the displacement of a target object in a first video frame and a second video frame of the target video is greater than or equal to a preset value, decoding the video frame data to obtain the first video frame and the second video frame, and inserting a third video frame according to a preset rule; and playing the first video frame, the second video frame and the third video frame.

The invention discloses an earphone with an automatic volume adjustment function, which comprises: an earphone main body and a sound outlet part, wherein the earphone main body comprises a front cavity MIC close to the sound outlet part so as to acquire an environment audio signal of the sound outlet part; a front cavity audio conversion circuit which is used for converting an environment audio signal acquired by the front cavity MIC to obtain a corresponding environment audio electric signal; a rear cavity MIC which is used for collecting an external real-time environment sound signal; a rearcavity audio conversion circuit which is used for converting an external real-time environment sound signal acquired by the rear cavity MIC to obtain a corresponding external real-time environment sound electric signal; and a processing control module which is used for analyzing and processing according to the environment audio electric signal and the external real-time environment sound electricsignal to obtain corresponding environment audio volume data and external real-time environment sound volume data, obtaining the required playing volume in combination with a pre-stored basic playingvolume data ratio, and adjusting and controlling the playing volume to be equal to the required playing volume. The invention further discloses a volume adjusting method for automatically adjusting the volume.