8 results about "Audio signal flow" patented technology

The invention relates to the technical field of wireless audio transmission, and discloses an audio sharing loudspeaker device matched with a heart sound stethoscope for teaching. The device comprises a parameter acquisition unit, a calibration unit, a clock unit, a distribution unit, a decoding unit, a feedback unit and an evaluation unit, wherein the parameter acquisition unit acquires environmental data and generates audio adjustment parameters; the calibration unit calibrates the teaching audio signal according to the audio adjustment parameter; the clock unit extracts time stamp information from the calibrated audio signal flow and realizes clock synchronization; the distribution unit generates and pushes multiple paths of audio distribution streams based on time synchronization, and monitors transmission interference and processes data packet loss at the same time; the decoding unit decodes the audio signal stream and adjusts the volume; the feedback unit collects audio playing effect feedback data and optimizes transmission; the evaluation unit evaluates audio distribution quality and reasonably switches channels. According to the invention, the problems of asynchronous playing and unstable sound quality in multi-terminal audio teaching are solved, and the synchronism and teaching quality of heart sound auscultation teaching are improved.

A noise cancellation technique is presented with improved frequency resolution. The method includes: acquiring a digitized noise signal from an environment in which the audio signal stream is present; receiving a data sample from the digitized noise signal; appending one or more additional samples to the data sample to form a series of samples, where magnitude for each of the one or more additional samples is substantially zero; computing a frequency domain representation of the series of samples in the frequency domain; shifting the frequency domain representation of the series of samples in time using the digital processor circuit, thereby producing a shifted frequency domain representation of the series of samples; converting the shifted frequency domain representation of the series of samples to time domain to form a portion of an anti-noise signal; and outputting the anti-noise signal into the audio signal stream to abate the noise through destructive interference.

The utility model discloses a two-way talkback processing system, and belongs to the field of voice signal processing. The system combines an advanced audio processing technology, and particularly uses an audio echo processing chip and an audio power amplifier module. Through an accurate echo cancellation algorithm (including frequency domain and time domain filtering), dynamic noise suppression and real-time gain adjustment, the system can significantly improve audio signal quality, avoid echo and self-excitation squeal phenomena, and ensure voice definition. The two-way audio signal stream design and the modular structure of the system enable the system to adapt to complex noise environments and various audio application scenarios, including talkback between drivers and passengers and processing of emergency alarm signals. In addition, through the design of audio bus transmission, audio power amplifier module amplification, transformer isolation and the like, stable transmission and high-quality output of audio signals are ensured. On the whole, the stability, definition and adaptability of the audio communication system are improved, and the echo problem in a traditional system is solved.

Systems, methods, and computer storage media are disclosed for transcript aggregation. In an example implementation, an audio mix processing map is generated representing an audio signal flow defined by a multi-track sequence that mixes audio sources arranged in audio tracks into a final mix. Using the audio mix processing map, an aggregated audio track is generated that represents which of the audio sources contribute to the final mix during which ranges of time. The aggregated audio track is transformed into an aggregated transcript track by associating transcript segments from transcripts of the audio sources with corresponding ranges of the ranges of time.

This invention relates to the field of audio control technology, and more particularly to an adaptive audio control method and system for in-vehicle scenarios. The method includes the following steps: acquiring in-vehicle audio signal streams based on an in-vehicle microphone array, performing dynamic range compression and frequency response balance optimization to construct an in-vehicle reference audio stream; analyzing the acoustic characteristics at multiple locations based on the in-vehicle reference audio stream, and performing acoustic transmission path distribution evolution to construct a three-dimensional sound field model of the vehicle cabin; extracting multi-dimensional driving behavior data, performing dynamic driving preference modeling, and constructing a driving behavior profile; calculating audio sensitivity based on the driving behavior profile, and dynamically adaptively adjusting it based on the three-dimensional sound field model of the vehicle cabin to construct dynamic audio adjustment parameters; applying safety optimization constraints and adaptive sound quality gain to the dynamic audio adjustment parameters, and outputting the final in-vehicle audio. This invention adjusts audio parameters in real time based on the vehicle scenario, improving driving safety and auditory experience.

This invention provides a multi-source automatic gain control method, system, electronic device, and medium, belonging to the field of vehicle technology. The method includes: acquiring raw audio data from multiple audio sources; converting the raw audio data into an audio signal stream; analyzing the signal energy of the audio signal stream to obtain a current signal energy value; obtaining a gain adjustment coefficient based on the current signal energy value and a preset target level value; smoothing the gain adjustment coefficient to obtain a target gain control value; adjusting the gain of a digital signal processor according to the target gain control value; and outputting an audio signal with a target loudness. This invention eliminates the problem of sudden volume changes caused by inconsistent output levels of different audio sources by dynamically adjusting the gain in real time based on the signal energy value (RMS value) of the audio signal stream, thus maintaining a stable and consistent output loudness. It utilizes RMS value calculation, which is computationally efficient and has low resource overhead, as the core of its dynamic gain control, which operates in real time and does not significantly increase the system load.

Communications system used for trading communications that interfaces with a transcription system to enable a service of transcribing trader communications. During an audio call, a stream of audio signals transmitted by a communications device via a user-selected communications line are received together with metadata from the communications device. The audio signals and metadata can be provided during the audio call and via an API gateway to a transcription system in one or more data formats that enable the transcription system to transcribe the stream of audio signals into text and tag the text with the metadata.

The invention relates to a speech translation method and device, equipment and a medium in the technical field of the Internet, and the method comprises the steps: responding to a speech translation event, continuously receiving an original audio signal flow corresponding to the event, and determining a plurality of voice-containing frame sequences corresponding to the original audio signal flow; for each human voice-containing frame sequence, performing non-human voice noise reduction processing on the human voice-containing frame sequence to obtain a pure human voice frame sequence; reasoning a to-be-translated text sequence corresponding to the pure human voice frame sequence by adopting a voice recognition model; and translating the to-be-translated text sequence into a target text sequence according to a target language corresponding to the speech translation event. The method can efficiently and accurately translate the voice in a streaming manner, and greatly improves the translation robustness in the acoustic environment of noise interference and mixed sound sources.