Digital Audio Stream Rate Adaptation to Match Playback Capabilities
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Solution Overview
Problem
Audio playback systems face limitations in dynamically adjusting the data rate of digital audio streams based on their capabilities, leading to suboptimal sound quality and bandwidth usage due to fixed or limited audio processing capabilities.
Innovation Solution
A method and system that sample sound generated by the audio playback system, analyze the quality, and adjust the data rate by reducing the sampling rate and bits per sample, using a computing device to determine the maximum usable sampling rate and bit depth, allowing for bandwidth conservation with minimal sound quality loss.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Loss of energy
If the data rate of the digital audio stream is reduced to conserve bandwidth, then bandwidth usage is improved, but sound quality deteriorates
Solution Approach 1:
The patent implements dynamic adjustment of audio stream parameters (sampling rate, bit depth, channels) based on real-time detection of playback system capabilities. The system transitions from static, predetermined audio rendering to dynamic adaptation, continuously monitoring the playback chain and adjusting the digital audio stream parameters accordingly to optimize both bandwidth efficiency and sound quality.
Solution Approach 2:
The system changes multiple parameters of the digital audio stream including sampling rate (e.g., 44.1kHz to 22.05kHz), bit depth (e.g., 16-bit to 12-bit), and channel configuration (e.g., stereo to mono) based on detected playback system capabilities. These parameter changes enable bandwidth reduction while maintaining acceptable sound quality for the specific playback environment.
2Device complexity
If fixed or limited audio processing capabilities are used in the audio playback system, then device complexity is reduced, but adaptability to different audio stream qualities deteriorates
Solution Approach 1:
The patent implements a feedback mechanism where the system detects the actual playback chain characteristics (amplifier, speakers, room acoustics) and uses this information to adjust the digital audio stream parameters. The detection process analyzes the playback system's response to test signals and feeds this information back to the audio source device, enabling automatic adaptation without requiring complex predetermined configurations.
Solution Approach 2:
The system performs self-diagnosis and self-adjustment by automatically detecting playback system capabilities and configuring the audio stream accordingly. The detection process and parameter adjustment occur without user intervention, with the system serving itself to optimize audio delivery based on the actual playback environment.
3Manufacturing precision
If high data rate audio streams are transmitted to maintain sound quality, then sound quality is improved, but bandwidth consumption increases
Solution Approach 1:
The system applies partial action by transmitting only the necessary audio information required for the specific playback system capabilities. Instead of always transmitting full-resolution high-data-rate audio, the system transmits reduced-resolution audio (lower sampling rate, bit depth, or channels) when the playback system cannot utilize the full quality, thereby reducing bandwidth consumption while maintaining perceptually adequate sound quality.
Data Source
AI summary
A computing device may be configured to output a digital audio stream to an audio playback system for rendering as sound over speakers. The sound may be sampled. Based at least in part on a quality of the sampled sound, the data rate of the digital audio stream may be reduced by reducing a sampling rate and/or by reducing a number of bits per sample. A reduced sampling rate may be determined based on a computed maximum sampling rate of the audio playback system, and/or a reduced number of bits per sample may be determined based on a computed maximum number of bits per sample of the audio playback system. The maximum usable sampling rate and maximum usable number of bits per sample may be determined based on an upper usable frequency within a frequency spectrum of the sampled sound.