Adaptive Audio Encoding Bit Rate for Consistent Speech Quality

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Solution Overview

Problem

Existing audio encoding methods using fixed encoding bit rates fail to ensure consistent quality due to the variability of speech signals over time, leading to inconsistent encoding quality.

Innovation Solution

An audio encoding method that dynamically adjusts the encoding bit rate based on audio feature parameters, such as pitch period and gain, using an encoding bit rate prediction model trained on various audio types to optimize encoding quality.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Reliability

If a fixed encoding bit rate is used, then the encoding process is simple and stable, but the encoding quality varies greatly for different speech signals

Engineering Contradiction:
Improveencoding quality consistencyVSAvoidencoding process complexity
Core Design Contradiction:
ReliabilityVSDevice complexity

Solution Approach 1:

The patent implements dynamic encoding bit rate adjustment by analyzing audio feature parameters (such as pitch period and gain) of each audio frame and selecting appropriate bit rates from a preset bit rate table. This transforms the static fixed bit rate system into a dynamic adaptive system that responds to varying speech signal characteristics, thereby ensuring consistent encoding quality across different speech signals while maintaining manageable system complexity through predefined bit rate options.

Inventive Principle:
Principle #15Dynamics

Solution Approach 2:

The patent changes the encoding bit rate parameter based on audio feature analysis. By computing features like pitch period and gain for each audio frame and mapping them to appropriate bit rates from a preset table, the system dynamically adjusts the encoding parameter to match speech signal characteristics, resolving the contradiction between quality consistency and process complexity.

Inventive Principle:
Principle #35Parameter changes

2Reliability

If a higher encoding bit rate is used, then the speech encoding quality is higher, but the bandwidth and storage space increase

Engineering Contradiction:
Improvespeech encoding qualityVSAvoidbandwidth and storage space
Core Design Contradiction:
ReliabilityVSQuantity of substance

Solution Approach 1:

The patent dynamically adjusts the encoding bit rate parameter based on audio feature analysis, selecting the minimum necessary bit rate from a preset table that achieves target quality. This avoids consistently using high bit rates, thereby reducing overall bandwidth and storage requirements while maintaining quality where needed.

Inventive Principle:
Principle #35Parameter changes

Solution Approach 2:

The patent applies encoding resources (bit rate) partially and adaptively rather than uniformly. By analyzing audio frame features and assigning bit rates only when and where needed based on speech characteristics, the system avoids excessive bit rate allocation for all frames, optimizing the balance between quality and resource consumption.

Inventive Principle:
Principle #16Partial or excessive action

Data Source

PatentEP4239630B1Audio encoding method, audio decoding method, apparatus, computer device, storage medium, and computer program product
Publication Date: 2026.04.22 TENCENT TECHNOLOGY (SHENZHEN) CO LTD
  • EP4239630B1 patent drawingFigure 1~3
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  • EP4239630B1 patent drawingFigure 5

AI summary

An audio encoding method, an audio decoding method, an apparatus, a computer device, a computer readable storage medium, and a computer program product, the audio encoding method comprising: acquiring a sample audio feature parameter corresponding to each sample audio frame in a first sample audio (401); performing encoding code rate prediction on the sample audio feature parameters by means of an encoding code rate prediction model, so as to obtain a sample encoding code rate of the sample audio frame (402); performing audio encoding on the sample audio frame on the basis of the sample encoding code rate, and generating sample audio data on the basis of an encoding result corresponding to each sample audio frame (403); performing audio decoding on the sample audio data to obtain a second sample audio corresponding to the sample audio data (404 ); and training the encoding code rate prediction model on the basis of the first sample audio and the second sample audio until a sample encoding quality score reaches a target encoding quality score to end the training (405), wherein the sample encoding quality score is determined by means of the first sample audio and the second sample audio.