Signal Processing Modules
The Loudness module provides for the analysis of audio Loudness, PPM and True Peak, together with correction if required. Loudness measurements available depend on the chosen specification, and Program Loudness, Short-Term Loudness and Momentary Loudness are supported. Dolby’s Dialog Intelligence algorithm is available as an additional option should this be required in conjunction with Program Loudness measurements.
The module also includes the measurement of LRA, and simple compression to reduce the LRA to a target. Reports are available of all measured parameters and can be produced in XML, PDF and CSV format. Graphs of Loudness, PPM and True Peak may be embedded into PDF reports.
Loudness is measured to:
- EBU R128
- ATSC A/85
- AGCOM 219/09/CSP
PPM is measured to:
- dBFS Type I
- dBFS Type II
True Peak is measured to:
LRA is measured to:
- EBU Tech 3342
The Korean Loudness standard is also supported, which uses a standard measurement algorithm but has additional reporting requirements.
Measurement/correction can be done for mono tracks, stereo tracks, and for 5.1 tracks. Any number of audio channels can be measured and corrected in a single pass, up to a maximum of 64 channels.
Where tracks contain Dolby E, these can be measured and corrected if the Dolby E Loudness option is also purchased.
Similarly, where tracks contain Dolby Digital or Dolby Digital Plus, these can be measured and corrected if the relevant option is purchased.
For all of Dolby E, Dolby Digital and Dolby Digital Plus, the pair of channels containing the encoded audio should be defined as a stereo pair in the workflow, and Engine automatically expands the audio based upon the contents of the Dolby, processes for loudness, then reencodes, maintaining the original AC3 metadata, but with a new Dialnorm value.
We have recently updated Engine to support loudness measurement and correction of MPEG4 files with AAC audio. In addition, we now also support loudness measurement and correction for PCM based immersive and ATMOS formats.
Upmix & Downmix
Upmix and Downmix The Upmix and Downmix modules lets you Upmix a stereo to 5.1, or to Downmix a 5.1 to stereo. The Upmix and Downmix technology are licensed from Soundfield.
The Soundfield Upmix splits the source stereo into direct sound and ambient sound, and steers and filters these separately prior to creating the 5.1 output. Standard presets are provided for News, Sports, Movies, Pop music and Classical music, or custom presets can be created. When creating a custom preset, the following controls are available.
- Direct Sound level
- Front Ambient level
- Rear Ambient Level
- Centre Divergence
Custom presets are stored for future use.
The Soundfield Downmix module creates stereo from a 5.1 source. Standard presets are provided, and custom presets can be created. When creating a custom preset, the following controls are available.
- Lt/Rt (Matrix on)
- Lo/Ro (Matrix off)
- LFE On/Off
- Centre Gain
- Ls/Rs Gain
The Stereo to Mono downmix function converts an ordinary stereo signal in to a mono signal, by doing a simple mix-down.
Channel Mapping and Mute
The Channel Mapping and Mute module provides for the manipulation of audio tracks within any supported file format, which includes MXF, QT/MOV, LXF, GXF, WAV and AIFF.
Channel Mapping can be used in combination with media files containing up to 64 tracks of audio.
- Any track can be duplicated to any other track location
- Any pair of tracks can be swapped
- Any track can be muted (set to silence for the entire duration of the media)
Tracks may be manipulated in groups in any combination
Defining tracks into groups has no functional effect on the file, but is used so simplify the remapping or muting configuration.
This option adds the ability alter either the pitch or the duration of the audio
Conversion of material between US and European frame rates is commonly done in files by restamping the frame rate. This causes the video to be played faster or slower than in the original, but the audio playout is fixed at 48 KHz so the audio is completely unchanged.
The purpose of this module is to change the duration of the audio to match that implied by the new video frame rate, whilst leaving the pitch of the audio unchanged from that in the source.
The module is controlled by the user specifying the source and destination frame rates used for the video, and it then calculates and applies the required audio duration adjustment.
Audio Description Module
Audio Description (also called Descriptive Video Service (DVS) is service to make video content more accessible to visually impaired people.
The Admix module within Engine provides a simple method for mixing the program audio with AD commentary track. This function is particularly useful for delivering VOD content, and for delivery to mobile devices, where hardware for AD merging is not normally present.
MXF with program audio AD + Control from external WAV file: The Audio Descriptor module takes Program Audio, Mono Audio descriptor Audio and uses the Control track to creates a new combined Audio Mix.
MXF file has program audio as well as Audio Descriptor + Control.
LRA (Loudness Range Processor)
LRA is a parameter defined in the EBU Tech 3342 Loudness specification. Most applications have a flexible guideline as to the maximum permitted LRA. A few countries, and specifically Brazil, set an absolute maximum value for LRA.
Loudness range is a generic measure that helps to decide if Dynamic Compression is needed.
This option in the Loudness module, lets you specify the maximum LRA for the output file to meet the requirement. As a compression algorithm is used, this option should only be enabled when required, as compression in an automated environment will change the overall mix of the sound.
The Audio Processor module uses a unique algorithm that converts the dynamics of original movie mixes so they are suitable for broadcast on terrestrial television or converts television mixes to be more suited to online broadcast. This is done very differently to traditional audio compressors that leave the sound ‘flat’ and with low dynamic waveforms. Our unique algorithm gives astonishingly good results with a wide range of program material.
The module has been designed with a very simple interface. Basically, for each audio track within the media, you select whether you want the output sound to be one of Open, Mid, Tight, or Very Tight.
On the whole, the Open and Mid settings will give results suitable for broadcast television, and the Tight and Very Tight settings give results suitable for online delivery. However, this is very dependent on your particular requirements.
The Audio Alignment module lets you adjust the position of the audio in the file, without changing the position of the video. Therefore the audio is moved earlier to later relative to the video, and this can be used to correct lipsync style offsets that may already exist in the content, or to add in a specific offset in advance, knowing that downstream processing is going to make a further change, and in this way end up with perfectly synchronised content.
From the dropdown, choose whether you are calculating the required alignment in frames, or in audio samples. If in frames you will also need to specify the frame rate. Then specify your track as one of mono, stereo or 5.1, and then select from the dropdown to either advance or delay the audio. Next, enter the number of frames or samples adjustment that you need, and click Add++.
The maximum adjustment available is 30 frames.
You need only specify the adjustment needed for required tracks, and the unspecified audio channels will retain their original position on the timeline.
Once adjustments for all required tracks are entered, click Save to store these and close the dialog.
Mono to Stereo
The mono to stereo module lets the user create a stereo mix out of a single mono stream, this will result in a 2 channel output file which has stereo characteristics. This module has been designed for broadcasters dealing with catalogues of archived content. Archived footage typically has mono audio, however, mono audio is not sufficient for modern-day play-out with most mono and dual mono files being rejected at QC. Using the mono to stereo module will allow archived content to comfortably pass through QC and move on to play-out.
Speaker Angle - This is automatically set to 45° as mathematically, this is the ideal speaker angle/separation. In reality, lots of studios have their speakers set 30° to 40° angled to the listener (quite often 38°). So this option is adjustable.
The speaker angle creates distance between the speakers, without this distance you are effectively playing the audio out as mono. By increasing the distance between the 2 speakers, you allow for each ear to hear the different information from speaker L and speaker R and therefore create a stereo effect. The lowest you can set the speaker angle to is 10°.
Stereo Width - This refers to the perceived size of the stereo image.
Control of the stereo width is provided in the settings tab. This will directly affect the perceivable width of the audio that you are processing. Values for this control are automatically set to 6 to start with. However, this can be adjusted so that the stereo image can be narrower (3) or wider (10). A greater stereo width allows for more spread in the stereo field. Here we can see 2 examples:
- Figure 1 shows a Signal which has a stereo width set to 0. This provides a mono signal and therefore is not useful for turning our mono files into stereo.
- Figure 2 shows a signal that has a stereo width set between 3 and 10. Here we can see a lot more spread in our stereo image.