Broadcast Level Corrector is a DSP Matlab function that processes an audio input wave file by applying ITU 1770 level correction and peak limiting.Architecture & Allied Art

Beverley, Benjamin John

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DESC9115: Digital Audio Systems – Lab Report 2 Ben Beverley – SID: 420170037 1  DESC9115: DIGITAL AUDIO SYSTEMS 2013 LAB REPORT 2 Instructor: William L. Martens Student: Ben Beverley SID: 420170037 Date: Tuesday 28th May, 2013   DSP Application: Broadcast Loudness Corrector  Note: This report refers to the following attached Matlab files: broadcast_level_corrector.m, CALL_broadcast_level_corrector.m, loudness_itu.m  In January 2013, new broadcast audio delivery standards that had already been implemented in the USA and Europe came into effect in Australia. The new standards, outlined in the Free TV Australia Operational Practice OP-59 (2010), are based on the International Telecommunication Union recommendation ITU-R BS.1770-3 (2012). The ITU recommendation determines methods and specifications of audio metering to ensure TV viewers experience a more even loudness of content across different programmes and commercials.   There has been a number of digital audio applications released to aid in accurate loudness monitoring by audio professionals. These plugins and standalone applications can be time consuming, expensive and difficult to operate, particularly for non-audio professionals. It is common for video editors with limited knowledge of audio processing and metering to be required to deliver content direct to broadcast. Television networks too must implement “tech checks” on all delivered content, often rejecting deliveries based on inappropriate audio and loudness levels or even digital clipping. And at the consumer level, there is a distinct lack of any audio standards at all for amateur filmmakers providing videos to websites such as YouTube and Vimeo.  The Broadcast Loudness Corrector (BLC) is a DSP application that analyses, processes and exports single or multi-channel wave files to the latest audio broadcast specifications in ITU-R BS.1770-3. Through the use of true-peak measurement, a built-in limiter and automatic LKFS loudness matching, the BLC can accurately ensure that audio deliverables match loudness criteria with merely a few clicks.   Figure 1. Broadcast Loudness Corrector signal flow  DESC9115: Digital Audio Systems – Lab Report 2 Ben Beverley – SID: 420170037 2 Digital Signal Processing:  The BLC processor follows the signal flow seen in Fig. 1. The signal’s input levels are first assessed. This Level analysis occurs multiple times throughout the processing, as it is used to provide onscreen feedback to the user, as well as provide data for graph plotting. The peak levels are analysed by finding the maximum value of the following equation:  dB = 20 log10 x2  (1) where x  is the discrete data signal  A clipping detector (Abdulla, 2007) is implemented through a process that seeks out whether any same sample value occurs ten times in a row. If clipping is detected in the input signal, a warning box will appear.    Figure 2. Input clipping warning box !The ITU 1770 Loudness matching calls an external function ‘loudness_itu.m’ (Marui, 2012), which calculates the current LKFS value of the signal using equations and constants outlined in ITU-R BS.1770-3. Fig. 3 is a block diagram showing the signal flow of this LKFS loudness matching function for up to a 5.1 channel audio file (please note: only mono and stereo processing is currently supported the BLC function).   Figure 3. Block diagram of multichannel loudness algorithm (ITU-R BS.1770-3) !In the Matlab implementation, firstly the constants are set:    The signal is then downsampled back to 48kHz using Matlab’s ‘resample.m’ function, since this is the sampling frequency for which the ITU-R filter coefficients are defined. The channels are weighted according to the ITU-R specifications: the blockSize = 400;   % in ms overlapSize = 0.75;   % in percentage channelWeights = [1 1 1 0 sqrt(2) sqrt(2)];   % in L, R, C, LFE, Ls, Rs order absoluteThreshold = -70;   % in dB relativeThreshold = -10;   % in dB DESC9115: Digital Audio Systems – Lab Report 2 Ben Beverley – SID: 420170037 3 Left, Centre and Right are all unweighted (1), Left Surround and Right Surround are weighted by 2 , and the LFE is excluded (0).  A 2-stage pre-filtering is next applied using ITU-R specified FIR coefficients. The first stage of the K-filter is a shelving filter, which accounts for the acoustic effects of the head, whilst the second stage is a simple high-pass filter that serves to negatively weight the lower frequencies, somewhat matching the frequency response of the human auditory system. The mean square is then calculated by the following code:  The code uses a “for” loop to measure the mean square value for separate time intervals determined by the variable ‘blocksize’. The final calculation for loudness over the measurement time (LK) is calculated:   LK = −0.691+10 log10 Gii∑ ⋅ zi  (2) where Gi are the weighting coefficients for each channel and zi is the mean square of the filtered input signal for a measured time interval (ITU-R BS.1770-3)  A two-stage process is then used to make the gated measurement:  Now that the signal has been analysed for apparent loudness, a brickwall feed-forward peak limiter (Zölzer, 2002) is applied. The limiter itself is designed to hard-limit any signal exceeding the limiter threshold (LT). As the plugin is designed for broadcast and the levels will have already been adjusted to reach the LKFS target (default -24dB), it is not likely the limiter will be required apart from to tame the occasional transient peak. The recommended maximum peak for broadcast deliveries in Australia is -2dBFS, which ensures there is a small amount of headroom to account for any inaccuracies in metering. This is the default value for the limiter if the user defines no other amount.  for n1 = 1:length(j)   yyy = yy(blockSize*j(n1)*(1-overlapSize)+1:blockSize*(j(n1)*(1-overlapSize)+1)+1,:);   z(n1,:) = sum(yyy .^ 2) / blockSize; end loudness_K = -0.691 + 10*log10(sum(repmat(chwat,length(z),1) .* z, 2)); % Gating (absolute) -  Jg = loudness_K > absoluteThreshold; % when loudness exceeds absolute threshold zz = repmat(channelWeights(1:numch),length(z),1) .* z; % replicates and tile arrays zeros according to number of channels L_KG = -0.691 + 10*log10(chwat * sum(zz(Jg,:), 1)' / sum(Jg)); % calculation for gated loudness amount Gamma_r = L_KG + relativeThreshold; % sum of gated loudness and relative threshold Jg = loudness_K > Gamma_r; % represents when loudness exceeds relative thresh L_KG = -0.691 + 10*log10(chwat * sum(zz(Jg,:), 1)' / sum(Jg)); % calculation for relative loudness amount DESC9115: Digital Audio Systems – Lab Report 2 Ben Beverley – SID: 420170037 4   AT = Attack Time RT = Release Time LT = Limiter Threshold LS = Limiter Slope Figure 4. Block diagram of a limiter The limiter’s signal processing, seen in Fig. 4, is a feed-forward design in which the input signal x(n) is split into a delayed component x(n-D) and a side chain path. The side-chain processing is done using the following code:  Firstly the variable (n) is defined by the input (x_lkfs1). The absolute value of the signal is found and then delayed by 1. The peak detector algorithm, which is dependent on whether the limiter is in attack (AT) or release (RT) mode, is then applied. If the signal is greater than the threshold (LT) then the limiter processing is performed to determine the gain factor (f). The processing of the gain factor is the linear calculation of the following equation:   f (n) =10−LS (X−LT )  (3) where X is the logarithmic value of x (Zölzer, 2002)  The final calculation of the limiter is the multiplication of the input signal by the gain factor. The processing is then repeated if a second channel is present.  If any limiting is applied then the overall LKFS value of the signal will have changed; meaning that the signal must undergo another level match to ensure the target LKFS value is met. This second matching process only runs if the analysis determines that the limiter input signal’s peaks are greater than the limiter output.        for n=2:length(x_lkfs1)    a=abs(x_lkfs1(n,1))-x_peak(n-1); % delays signal by n-1    if a<0, a=0; end;    x_peak(n) = x_peak(n-1)*(1-RT) + AT*a; % Peak detector depending on AT/RT    if x_peak(n)>lt, % if the signal exceeds the threshold...        % ...run the limiter processor:       f_1(n)=10^(-LS*(log10(x_peak(n))-log10(lt))); % linear calculation of…    else f_1(n)=1; % ...otherwise let the gain factor remain as 1    end; end; limit_out(:,1)=x_lkfs1(:,1).*f_1; % apply gain adjustment DESC9115: Digital Audio Systems – Lab Report 2 Ben Beverley – SID: 420170037 5 Further Development:  There is further room for development of this processor in the future: the first and most obvious being the design of a graphical user interface (GUI) containing various preset settings that ensure the user experience is both fast and intuitive.   Support for different file formats could be easily implemented, along with better support for surround sound files. Up to 5.1 channels processing is in fact already implemented in the ‘loudness_itu.m’ function. However testing revealed that Matlab’s ‘wavread.m’ function does not currently support more than 2 channels of audio; an issue that could be easily overcome with more development.  With further research, a more transparent limiter could be introduced that has intelligent auto-release for less audible transient control. This feature is not too necessary for the current application, but would be ideal if the processor were to be developed for real-time analysis in DAWs and not just post-export corrections.                                   DESC9115: Digital Audio Systems – Lab Report 2 Ben Beverley – SID: 420170037 6 References:  Abdulla, K., 2007. Matlab Central: Clipping Detection. The Mathworks Inc. [ONLINE] Available at: http://www.mathworks.com.au/matlabcentral/newsreader/view_thread/159887.  [Accessed 15 May 13].  Free TV Australia 2010, Operational Practice OP-59: Measurement and Management of Loudness in Soundtracks for Television Broadcasting, Issue 1, Sydney.  International Telecommunication Union 2012, Recommendation ITU-R BS.1770-3: Requirements for loudness and true-peak indicating meters, International Telecommunication Union, Geneva.  International Telecommunication Union 2012, Recommendation ITU-R BS.1771-1: Algorithms to measure audio programme loudness and true-peak audio level, International Telecommunication Union, Geneva.  Marui, A., 2012. Loudness calculation based on Rec. ITU-R BS.1770-2. [ONLINE] Available at: http://www.geidai.ac.jp/~marui/matlab/node42.html [Accessed 22 April 13]  Zölzer, U, 2002. DAFX - Digital Audio Effects. 1st ed. West Sussex, England: John Wiley & Sons, Ltd.  Zölzer, U, 2008. Digital Audio Signal Processing. 2nd ed. West Sussex, England: John Wiley & Sons, Ltd.DESC9115: Digital Audio Systems – Lab Report 2 Ben Beverley – SID: 420170037 7 Appendix: Please find the following supplementary audio files attached:  Input Files Channels Output Files Broadcast Level Corrector Settings Target LKFS (dB) Limiter Threshold (dB) input_eg1_film_stereo.wav Stereo output_eg1_film_stereo.wav -24 -2 input_eg2_music_mono.wav Mono output_eg2a_music_mono.wav -24 -2 output_eg2b_music_mono.wav -24 -10 Table 2. Input audio files vs. output audio files and the BLC settings used during processing        Input file: input_eg2_music_mono.wav  Figure 5. Plotted graph showing the processing of a loud mono music input signal. The lines across the audio show the LKFS level of the input and output files. As is evident from the limiter graph, no limiting incurred. This file did however have clipping in the raw data, prompting the Clipping Warning in Matlab.  Input file: input_eg2_music_mono.wav  Figure 6. The same input as Fig. 6 is shown here with the target loudness level still -24 dB (LKFS), but the limiter threshold set to -10dBFS. DESC9115: Digital Audio Systems – Lab Report 2 Ben Beverley – SID: 420170037 8    input_eg1_film_stereo.wav Figure 7. Plotted graphs showing a stereo audio signal pre-processing and post-processing. The input signal was a 30 second snippet from a short film. The bottom graph shows the Limiter Output Signal Gain. This signal had a distinct peak toward the middle, which after the LKFS gain increase would have resulted in clipping if the limiter were not present. 

Broadcast Level Corrector

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Broadcast Level Corrector

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