Object-Based Audio Loudness Compensation for Flexible Rendering
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Solution Overview
Problem
Conventional audio rendering technologies face challenges in maintaining consistent loudness across different audio programs and channels, leading to annoying volume differences during switching, as they lack accurate and immediate loudness estimation and compensation mechanisms.
Innovation Solution
A decoder and encoder system that receives audio object signals and loudness information, determining a compensation value based on rendering and loudness data to generate output channels with consistent loudness, using parametric methods for accurate and immediate loudness estimation and compensation.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If audio rendering is conducted on the decoder side to allow flexible manipulation of audio objects, then adaptability and ease of operation are improved, but the loudness consistency deteriorates due to significant effects on overall/total loudness
Solution Approach 1:
The encoder pre-calculates and embeds loudness information (such as loudness levels, true peak levels, or other loudness parameters) into the bitstream alongside the audio objects. This preliminary action enables the decoder to predict the loudness impact of rendering operations without actually performing the rendering, allowing for proactive loudness compensation to maintain consistency while preserving rendering flexibility
Solution Approach 2:
The system implements a feedback mechanism where the decoder uses the pre-encoded loudness information to monitor and adjust the loudness of rendered output in real-time. By comparing the predicted loudness with target loudness levels, the decoder can apply compensatory gain adjustments to maintain loudness consistency across different rendering scenarios while preserving adaptability
2Stability of the object's composition
If conventional loudness normalization is applied to all programs to a common reference level, then loudness consistency between programs is improved, but the complexity of maintaining accurate loudness estimation increases
Solution Approach 1:
The patent extracts loudness-related parameters (such as integrated loudness, true peak, or momentary loudness values) from the full audio signal and encodes them as separate side information in the bitstream. This extraction simplifies the normalization process by providing ready-to-use loudness metrics that the decoder can directly apply for normalization without complex real-time analysis, reducing device complexity while maintaining accuracy
Solution Approach 2:
The system changes the representation of loudness information by encoding multiple loudness parameters (integrated loudness, true peak, momentary loudness) in different temporal and spectral domains. This parameter transformation enables efficient normalization operations and simplifies the complexity of maintaining accurate loudness estimation across diverse audio programs
Data Source
AI summary
A decoder for generating an audio output signal having one or more audio output channels includes a receiving interface for receiving an audio input signal including a plurality of audio object signals, for receiving loudness information on the audio object signals, and for receiving rendering information indicating whether one or more of the audio object signals shall be amplified or attenuated. Moreover, the decoder includes a signal processor for generating the one or more audio output channels of the audio output signal. The signal processor is configured to determine a loudness compensation value depending on the loudness information and depending on the rendering information. Furthermore, the signal processor is configured to generate the one or more audio output channels of the audio output signal from the audio input signal depending on the rendering information and depending on the loudness compensation value. Moreover, an encoder is provided.