Loudness Descriptors to Characterize Wide Loudness-Range Material

The recent loudness measurement recommendations by the ITU and the EBU have gained widespread recognition in the broadcast community. The material it deals with is usually full-range mastered audio content, and its applicability to multitrack material is not yet clear. In the present work we investigate how well the evaluated perception of single track loudness agrees with the measured value as defined by ITU-R BS.1770. We analyze the underlying features that may be the cause for this disparity and propose some parameter alterations that might yield better results for multitrack material with minimal modification to their rating of broadcast content. The best parameter sets are then evaluated by a panel of experts in terms of how well they produce an equal-loudness multitrack mix, and are shown to be significantly more successful.

2016

This article describes one of the most fundamental changes in the history of audio in broadcasting: the change of the levelling paradigm from peak normalisation to loudness normalisation. This change is vital because of a problem that has become a major source of irritation for television and radio audiences around the world – that of the jump in audio levels at the breaks within programmes, between programmes and between channels. Loudness normalisation is the solution to counteract this problem. EBU Recommendation R 128 [1] establishes a predictable and well-defined method to measure the loudness1 level for news, sports, advertisements, drama, music, promotions, films etc., through-out the broadcast chain, and thereby helps professionals to create robust specifications for ingest, production, playout and distribution to a multitude of platforms. Four supporting documents have been created by the EBU to aid the audio industry to work with Rec. R 128. It is based entirely on open st...

Audio Engineering Society Convention 134, 2013

Over the last few years, loudness control represents one of the most frequently investigated topics in audio signal processing. In this paper we describe a framework designed to provide adaptive real-time loudness measurement and processing of audio files and streamed content being reproduced by mobile players hosted in laptops, tablets, and mobile phones. The proposed method aims to improve the users’ listening experience by normalizing the loudness level of the content in real-time, while preserving the original creative intent of the original soundtrack. The loudness measurement and adaptation is based on a customization of the High Efficiency Loudness Model algorithm described in the AES Convention Paper #8612 (“HELM: High Efficiency Loudness Model for Broadcast Content,” presented at the 132nd Convention, April 2012). Technical and subjective tests were performed in order to evaluate the performance of the proposed method. In addition, the way the subjective test was arranged offered the opportunity to gather information on the preferred Target Level of streamed and media files reproduced on portable devices.

The Journal of the Acoustical Society of America, 2017

During the early years of the International Telecommunication Union (ITU) loudness calculation standard for sound broadcasting [ITU-R (2006), Rec. BS Series, 1770], the need for additional loudness descriptors to evaluate short-form content, such as commercials and live inserts, was identified. This work proposes a loudness control scheme to prevent loudness jumps, which can bother audiences. It employs shortform content audio detection and dynamic range processing methods for the maximum loudness level criteria. Detection is achieved by combining principal component analysis for dimensionality reduction and support vector machines for binary classification. Subsequent processing is based on short-term loudness integrators and Hilbert transformers. The performance was assessed using quality classification metrics and demonstrated through a loudness control example.

Archives of Acoustics, 2000

The present study was carried out to determine whether recorded musical tones played at various pitches on a clarinet, a flute, an oboe, and a trumpet are perceived as being equal in loudness when presented to listeners at the same A-weighted level. This psychophysical investigation showed systematic effects of both instrument type and pitch that could be related to spectral properties of the sounds under consideration. Level adjustments that were needed to equalize loudness well exceeded typical values of JNDs for signal level, thus confirming the insufficiency of A-weighting as a loudness predictor for musical sounds. Consequently, the use of elaborate computational prediction is stressed, in view of the necessity for thorough investigation of factors affecting the perception of loudness of musical sounds.

2014

2014

Recently, several researchers have been focusing on loudness perception of sound sources when the sounds are presented with the appearance of what radiates them. They concluded the visual information may increase/decrease the perceived loudness intensity. However, because these works concentrated mostly on subjective evaluation of actual sounds, there is a demand for an objective loudness model in consideration of the audio-visual interaction. Due to this reason, this paper focuses on its development. First of all, a new equal loudness curve considering the color effect is proposed for a limited frequency band based on experimental results of audio-visual stimuli consisting of sinusoidal sounds and color patches. Secondly, this function is applied to the loudness calculation to correct the effect of colors, whereas, the other computation processes are the same as a conventional loudness model of ISO532B. This is because it is considered that the mechanism of sound perception is not ...

2006

To facilitate better consistency between programs and stations, ITU, EBU and ARIB have investigated the standardization of broadcast loudness. This paper examines some consequences of a global loudness standard with regard to metering and control at the Ingest, Production and Transmission stages. Findings are reported from the latest research into mono, stereo and multichannel loudness measurement of real-world broadcast sounds. The improvements achieved by the new loudness models are quantified against previous level descriptors, such as, for example, PPM and Leq(A). Besides from reducing consumer annoyance with jumping levels, less engineering time needs being spent per audio stream. This too, is important because digital broadcast means a significant proliferation of the number of channels and the number of platforms. Each platform, such as TV, radio, internet, podcast, and other personal entertainment systems, has its own requirements for dynamic range, frequency range and speec...

The Journal of the Acoustical Society of America, 2013

The calculation of perceived loudness is an important factor in many applications such as the assessment of noise emissions. Generally, loudness of stationary sounds can be accurately predicted by existing models. For sounds with time-varying characteristics, however, there are still discrepancies between experimental data and model predictions, even with the most recent loudness models. This contribution presents a series of experiments in which loudness was measured in normal-hearing subjects with different types of realistic signals using an adaptive loudness matching procedure and categorical loudness scaling. The results of both methods indicate that loudness of speech-like signals is largely determined by the long-term spectrum, while other speech-related properties (particularly temporal modulations) play only a minor role. Loudness of speech appears to be quite robust towards even severe signal modifications, as long as the long-term spectrum is similar. In contrast, loudness of technical, strongly impulsive signals is considerably influenced by temporal modulations. For some of the signals, loudness could not be predicted by current models. Since the perceived loudness was underestimated by the models for some signals, but overestimated for other signals, a simple adjustment of the employed time constants in the temporal integration stage could not eliminate the discrepancies.

2020

The ITU-R BS.1770 multichannel loudness algorithm performs a sum of channel energies with weighting coefficients based on azimuth and elevation angles of arrival of the audio signal. In its current version, these coefficients were estimated based on binaural summation gains and not on subjective directional loudness. Also, the algorithm lacks directional weights for wider elevation angles (jfj � 30�). A listening test with broadband stimuli was conducted to collect subjective data on directional effects. The results were used to calculate a new set of directional weights. A modified version of the loudness algorithm with these estimated weights was tested against its benchmark using the collected data, and using program material rendered to reproduction systems with different loudspeaker configurations. The modified algorithm performed better than the benchmark, particularly with reproduction systems with more loudspeakers positioned out of the horizontal plane.