856 results about "Audio mixing" patented technology

A visual audio mixing system which includes an audio input engine configured to input one or more audio files each associated with a channel. A shape engine is responsive to the audio input engine and is configured to create a unique visual image of a definable shape and/or color for each of the one or more of audio files. A visual display engine is responsive to the shape engine and is configured to display each visual image. A shape select engine is responsive to the visual display engine and is configured to provide selection of one or more visual images. The system includes a two-dimensional workspace. A coordinate engine is responsive to the shape select engine and is configured to instantiate selected visual images in the two-dimensional workspace. A mix engine is responsive to coordinate engine and is configured to mix the visual images instantiated in the two-dimensional workspace such that user provided movement of one or more of the visual images in one direction represents volume and user provided movement in another direction represents pan to provide a visual and audio representation of each audio file and its associated channel.

Teleconferencing and, in particular, distributed teleconferencing may use methods and systems for location grouping to reduce feedback and other audio anomalies. Terminals and users connected to the same teleconference and in the same location might not need to receive audio signals from the other terminals and users in the same location. As such, by detecting and analyzing the location of each participating terminal, the terminals (and thus, the users thereof) may be organized into location groups to provide proper audio mixing. In one example, first and second terminals in the same location might not receive each other's audio in a downstream teleconference signal. The location and grouping of terminals may be processed using context fingerprint information derived from sensor readings of each terminal. Sensors may include GPS sensors, cameras, BLUETOOTH sensors and the like. Context fingerprint information may further be synchronized to enhance location determination and grouping.

This invention is directed to a system and method for the close coupling of Session Initiation Protocol (SIP) based Voice over Internet Protocol (VoIP) with multiplayer network games. Game state dependent voice communications are used as another method of player interaction in the game environment. The system uses SIP enhanced game servers and conference servers with audio mixing capabilities, and the game server controls the creation, maintenance and changing of the voice conferences. Participation in a voice conference is determined by the game state features associated with each player including distance and direction to determine how the voices are mixed in a voice conference.

A digital audio mixing system for live performance venues includes a software user interface and system host PC with an internal digital signal processor to perform digital mixing functions. The system includes a console having an array of multiple touch screen displays with corresponding fader board (tactile) control surfaces operatively connected to the host PC, and an audio patch bay unit. One or more stage boxes are linked to each other and to the system host PC by wired or wireless connections. The user interface includes multiple functional views and configuration presets, displayed in setup and real time modes, to allow the user to operate the system in a user friendly and simplified environment.

The invention discloses a karaoke in-pair singing method based on an internet and a system thereof. A vocal accompaniment user terminal plays instrument accompaniment of a song as well as reverberation and audio mixing processing is carried out on the instrument accompaniment of the song and a part sung by the vocal accompaniment user, so that an audio rack with the instrument accompaniment and the vocal accompaniment of the song is formed; the audio rack with the instrument accompaniment and the vocal accompaniment of the song is forwarded to a chorus user terminal through a transfer server; the chorus user terminal carries out decoding on the audio rack with the instrument accompaniment and the vocal accompaniment of the song, plays the audio track and carries out reverberation and audio mixing processing on the audio rack with the instrument accompaniment and the vocal accompaniment of the song and a part sung by the chorus user, thereby forming an audio rack with the instrument accompaniment, the vocal accompaniment and the chorus of the song; group sending is carried out on the audio rack with the instrument accompaniment, the vocal accompaniment and the chorus of the song to a listening client by a group sending server; and listening is carried out on the client as well as decoding is carried out on the audio rack with the instrument accompaniment, the vocal accompaniment and the chorus of the song to carry out playing. In addition, the invention also discloses a karaoke in-pair singing system based on the internet. According to the invention, an ideal in-pair singing effect can be provided for a user in a complicated environment with a large delay.

A multichannel audio format provides a truly discrete as well as a backward compatible mix for surround-sound, front or other discrete audio channels in cinema, home theater, or music environments. The additional discrete audio signals are mixed with the existing discrete audio channels into a predetermined format such as the 5.1 audio format. In addition these additional discrete audio channels are encoded and appended to the predetermined format as extension bits in the bitstream. The existing base of multichannel decoders can be used in combination with a mix decoder to reproduce truly discrete N.1 multichannel audio.

The present invention is directed to a system, including apparatus and architecture, and method for providing scheduled and ad hoc voice over IP conference services having enhanced quality and reliability. The system uses an overlay network of geographically distributed conference servers working simultaneously and collaboratively to host a given conference session. Multiple conference servers simultaneously and collaboratively perform the necessary audio mixing. The voice routing path is established and controlled within the overlay network and supports conference services across different conference service providers.

The embodiment of the invention discloses a live broadcast sound processing method and a mobile terminal. The live broadcast sound processing method comprises the steps that an audio file selected by a user is acquired; the audio file is decoded to obtain an audio signal; a speech signal of the user is collected; the audio signal and the speech signal are subjected to audio mixing and broadcast. According to the live broadcast sound processing method and the mobile terminal, the live broadcast atmosphere can be enriched without needing other loudspeaking equipment, and the quality of a finally-output signal can be improved.

The invention discloses an audio mixing control method and apparatus for Wince vehicle-mounted multimedia. The audio mixing control method comprises: A, when supplications are started, generating corresponding process IDs, and when the already started applications need to play sound, creating corresponding audio streams; B, pairing the audio stream of each application and the corresponding process IDs for marking, and identifying the types of sounding applications according to the process IDs; and C, according to preset sound priorities, controlling the volume of an application whose priority is the highest to be highest, and closing or reducing the sound of an application whose priority is low. According to the invention, the volume of the application whose priority is the highest can be controlled to be highest according to the sound priorities, and the sound of the application whose priority is low can be reduced or closed, and such an automatic sound adjusting mode reduces the influence of other unnecessary sound, so that the problem can be solved that a user cannot listen clearly due to simultaneous playing of multiple sound at the same volume, manual operation by the user is unnecessary, the use of the user is greatly facilitated, and driving is not affected.

A system, method, and computer program product for establishing multi-party VoIP conference audio calls in a distributed, peer-to-peer network where any number of nodes are able to arbitrarily and asynchronously start or stop producing audio output to be mixed into a single composite audio stream that is distributed to all nodes. A single distribution tree is used that has optimal communications characteristics to distribute the composite audio signal to all nodes. An audio mixing tree is established and maintained by adaptively and dynamically adding and merging intermediate mixing nodes operating between user nodes and the root of the single distribution tree. The intermediate mixing nodes and the root of the single distribution tree are all hosted, in an exemplary embodiment, on user nodes that are endpoints of the distribution tree.

During mixing of an N-channel input signal to generate an M-channel output signal, the mixed channel signals are equalized (e.g., amplified) to maintain the overall energy/loudness level of the output signal substantially equal to the overall energy/loudness level of the input signal. In one embodiment, the N input channel signals are converted to the frequency domain on a frame-by-frame basis, and the overall spectral loudness of the N-channel input signal is estimated. After mixing the spectral components for the N input channel signals (e.g., using weighted summation), the overall spectral loudness of the resulting M mixed channel signals is also estimated. A frequency-dependent gain factor, which is based on the two loudness estimates, is applied to the spectral components of the M mixed channel signals to generate M equalized mixed channel signals. The M-channel output signal is generated by converting the M equalized mixed channel signals to the time domain.

An audio mixing system is disclosed which makes it possible, when the transmission path that was disconnected is reconnected, to carry out processing in which the control parameters for setting the audio processing apparatus are transmitted to the audio processing apparatus from the audio control apparatus and processing with which visual confirmation that the connection has been made again is provided to the user. It has a detection means in which a specified signal is transmitted in the transmission path that is used for transmitting and the fact that the loop connection of the transmission path has been completed is detected by means of the reception by the transmission path used for reception of the signal that corresponds to the specified signal that has been transmitted, and a change instruction means in which the state of the control is changed in conformance with the operation of a specified operator and, based on the control state that has been instructed by said change, a signal to carry out the control instruction is transmitted on the transmission path used for transmitting, and a transmission means in which, in the case where the fact that the loop connection of the transmission path has been completed has been detected by the detection means, the signal to carry out the control instruction based on the most recent control state in accordance with the change instruction means is transmitted on the transmission path that is used for transmission.

The invention discloses an audio mixing method and equipment thereof, so that a defect of a present server can be overcome, wherein the defect is that there is limitation of many conditions during audio mixing processing. The disclosed audio mixing method comprises the following steps that: a server receives and decodes voice coded data in the best fit sampling rate, channel amount and coding mode sent by at least two source clients, wherein formats of the decoded data are unified as a PCM format; the server determines whether the sampling rates and sound channels of the voice coded data transmitted by all source clients are in accord with standards; if not, the voice data of the source clients are sampled again according to a standard sample rate, wherein the sample rates of the voice data are different from the standard sample rate; and meanwhile, the voice data of the source clients are expanded or merged according to a standard sound channel, wherein the sound channels of the voice data are different from the standard sound channel; the voice data, which have a unified sample rate and a unified sound channel, are aligned according to a time stamp; and audio mixing processing is carried out on the aligned voice data, and then the processed voice data are sent to a lodge client. The disclosed method provide in the invention can be applied widely to a VOIP multi-people meeting.

Systems and methods are disclosed for mixing of video streams in a collaborative mobile environment. In an embodiment, method for mobile video mixing includes receiving a plurality of video streams at a mobile device corresponding to one or more cameras capturing an event. The event may correspond to a live team-based sports or a public event. The method further includes choosing a group of video streams intended to be previewed at the mobile device. In a successive progression, the group of video streams are combined to generate a combined video streams such that the combined video signal provides a simultaneous preview of the group of video signals when displayed on the display screen of the mobile device. A user (e.g. video director) can select a video stream from the preview of group of video streams and send a request for a broadcast of the selected video stream.

A system for linking together audio signal producing, signal processing, and ear coupling devices comprising, in an embodiment: (1) an in-ear audio coupling device; (2) hearing aid electronics; (3) an external audio signal generating device; and (4) a module for audio mixing and enhancement.

The invention discloses an implementation method, system and server for an audio-video conference over the internet. The implementation method comprises the followings: A, a signaling server and a plurality of streaming media servers are arranged in a wide area network, the signaling server connects a conference terminal according to the connecting request of the conference terminal and allocates a corresponding streaming media server for the conference terminal; B, the corresponding streaming media server receives original audio-video data from the conference terminal and carries out audio mixing processing and video splicing processing to synthesize audio-video data; C, the corresponding streaming media server responds to the request of the conference terminal, and sends synthesized audio-video data or the original audio-video data to the conference terminal sending the request. According to the invention, data transmission quantity of a video conference can be effectively reduced, a large scale video conference can be achieved, and the problems of bandwidth bottleneck and incapability of getting through a gateway by a part local area network can be solved.

The invention discloses a method and device for processing audio data in a many-to-one projection screen system. When many-to-one screen projection is carried out, each electronic equipment can play own audio and/or audios of other electronic devices according to user requirements, so that the requirements of different users in a many-to-one screen projection scene are met. When the electronic equipment performs audio mixing processing on own audios and audios of other electronic equipment and then plays the audios, a user can hear the audios of all the electronic equipment at the same time. In the game scene, the user can feel the game scene of himself/herself and can also feel the game scenes of other users, and the user experience can be improved.

The invention relates to the field of audio information processing, and discloses a multi-microphone audio mixing method and device, which can reduce the misjudgment rate of gating of input channels and improve the audio quality after audio mixing. The method comprises the following steps: counting the signal intensity of each input channel at the current time, selecting out at least two input channels with the maximum signal intensities for carrying out voice detection; determining the detected input channels with voice as voice input channels; when at least two voice input channels are determined, determining the similarity of signals between the voice input channels; controlling the gating of the voice input channels according to the similarities of the signals; and carrying out weighted audio mixing on the signals of the gated voice input channels and outputting.