Steinberg Cubase 8 Manual
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Audio processing and functions Detect Silence 421 Detect Silence The Detect Silence function searches for silent sections in an event and either splits the event, removing the silent parts from the project, or creates regions corresponding to the non-silent sections. • To open the Detect Silence dialog, select one or several audio events in the Project window or the Audio Part Editor. On the Audio menu, open the Advanced submenu and select “Detect Silence”. If you select more than one event, the Detect Silence dialog allows you to process the selected events successively with individual settings or to apply the same settings to all selected events at once. The settings have the following functionality: Open Threshold When the audio level exceeds this value, the function “opens”, i. e. lets the sound pass. Audio material below the set level is detected as “silence”. Set this value low enough to open when a sound starts, but high enough to remove unwanted noise during “silent” sections. Close Threshold When the audio level drops below this value, the function “closes”, i. e. sounds below this level are detected as “silence”. This value cannot be higher than the Open Threshold value. Set this value high enough to remove unwanted noise during “silent” sections. Linked If this checkbox is activated, the Open and Close Threshold values are always set to the same value. Min. time open Determines the minimum time that the function will remain “open” after the audio level has exceeded the Open Threshold value. If the audio contains repeated short sounds, and you find that this results in too many short “open” sections, try raising this value.
Audio processing and functions Detect Silence 422 Min. time closed Determines the minimum time that the function will remain “closed” after the audio level has dropped below the Close Threshold value. Set this to a low value to avoid removing sounds. Pre-roll Allows you to cause the function to “open” slightly before the audio level exceeds the Open Threshold value. In other words, the start of each “open” section is moved to the left according to the time you set here. This is useful to avoid removing the attack of sounds. Post-roll Allows you to cause the function to “close” slightly after the audio level drops below the Close Threshold value. This is useful to avoid removing the natural decay of sounds. Add as Regions “Add as Regions” will create regions according to the non-silent sections. If you activate the “Add as Regions” option, you can specify a name for the regions in the Region Name field. In addition to the name, the regions will be numbered, starting with the number specified in the “Auto Number Start” field. Strip Silence “Strip Silence” will split the event at the beginning and end of each non-silent section, and remove the silent sections in between. Process all selected Events If you have selected more than one event, you can activate the “Process all selected Events” checkbox to apply the same settings to all selected events. Compute The audio event is analyzed, and the waveform display is redrawn to indicate which sections are considered “silent” according to your settings. Above the Compute button, the number of detected regions is displayed. Auto If you activate the Auto checkbox next to the Compute button, the audio event is analyzed (and the display is updated) automatically every time you change the settings in the Detection section of the dialog. Deactivate this option when you are working with very long files, as this process might take some time.
Audio processing and functions Detect Silence 423 Adjustments in the waveform display The upper part of the dialog displays a waveform image of the selected audio event. In case you have selected several audio events, the waveform of the event that you have selected first is shown. You can make the following adjustments: • With the zoom slider below the waveform to the right, zoom in and out on the waveform. You can also click in the waveform, keep the mouse button pressed, and move the mouse for zooming. Move the mouse down to zoom in and move it up to zoom out. • If you have zoomed in on the waveform, it may not be completely visible anymore. In this case, the scrollbar to the left of the zoom slider allows you to scroll through the waveform. You can also use the mouse wheel for scrolling through the waveform. • If the Linked option in the Detection section is deactivated, you can use the green square at the beginning and the red square at the end of the audio file to graphically adjust the Open and Close Threshold values (respectively). When “Linked” is activated, you can use either square to adjust both values. The Open and Close Threshold values in the Detection section reflect these changes. Making settings and processing The lower part of the Detect Silence dialog provides settings for the detection and processing of “silent” sections. PROCEDURE 1. Adjust the settings in the Detection section to the left. 2. Click the Compute button. The audio event is analyzed, and the waveform display is redrawn to indicate which sections are considered “silent” according to your settings. Above the Compute button, the number of detected regions is displayed. 3. Click “Preview” to listen to the result. The event is played back repeatedly in its entire length, but with the “closed” sections silenced. 4. Adjust the settings in the Detection section until you are satisfied with the result. 5. In the Output section, activate the “Add as Regions” or the “Strip Silence” option, or both.
Audio processing and functions The Spectrum Analyzer 424 6. Click the Process button. The event is split and/or regions are added. NOTE If you have selected more than one event and did not activate the “Process all selected Events” option in the Output section, the dialog opens again after processing, allowing you to make separate settings for the next event. The Spectrum Analyzer This function analyzes the selected audio, computes the average “spectrum” (level distribution over the frequency range) and displays this as a two-dimensional graph, with frequency on the x-axis and level on the y-axis. PROCEDURE 1. Make an audio selection (a clip, an event or a range selection). 2. Select “Spectrum Analyzer” from the Audio menu. A dialog with settings for the analysis appears. The default values give good results in most situations, but you can adjust the settings if you like: •Size in Samples The function divides the audio into “analysis blocks”, the size of which is set here. The larger this value, the higher the frequency resolution of the resulting spectrum. •Size of Overlap The overlap between each analysis block. •Window used Allows you to select which window type is used for the FFT (Fast Fourier Transform, the mathematical method used for computing the spectrum).
Audio processing and functions The Spectrum Analyzer 425 •Normalized Values When this is activated, the resulting level values are scaled, so that the highest level is displayed as “1” (0 dB). •From Stereo When analyzing stereo material, there is a pop-up menu with the following options: Mono mix – the stereo signal is mixed to mono before analyzing. Mono left/right – the left or right channel signal is used for analysis. Stereo – both channels are analyzed (two separate spectrums will be displayed). 3. Click the Process button. The spectrum is computed and displayed as a graph. 4. You can adjust the display with the settings in the display window: •dB When this is activated, the vertical axis shows dB values. When it is deactivated, values between 0 and 1 are shown. •Freq. log When this is activated, frequencies (on the horizontal axis) are displayed on a logarithmic scale. When it is deactivated, the frequency axis is linear. •Precision Indicates the frequency resolution of the graph. This value cannot be changed here, but is governed by the Size in Samples setting in the previous dialog. •Frequency/Note Allows you to select whether you want the frequencies to be displayed in Hertz or with note names. •Min. Sets the lowest frequency shown in the graph. •Max. Sets the highest frequency shown in the graph. By adjusting the Min and Max values, you can take a closer look at a smaller frequency range. •Active When this is activated, the next Spectrum Analysis will appear in the same window. When deactivated, new Spectrum Analysis results will appear in separate windows. 5. If you move the mouse pointer over the graph, a cross-hair cursor follows the graph curve and the display in the upper right corner shows the frequency/note and level at the current position. To compare the level between two frequencies, move the pointer to one of the frequencies, right-click once and move the pointer to the second frequency. The delta
Audio processing and functions Statistics 426 value (the difference in level between the current position and the right-click position) is displayed in the upper right corner (labeled “D”). • If you analyze stereo audio and selected the “Stereo” option in the first dialog, the graphs for the left and right channel are superimposed in the display, with the left channel graph in white and the right channel graph in yellow. The display in the upper right corner shows the values for the left channel – to see the right channel values, hold down [Shift]. An “L” or “R” is displayed to indicate which channel values are shown. 6. You can leave the window open or close it by clicking the “Close” button. If you leave it open and the “Active” checkbox is ticked, the result of the next Spectrum Analysis will be displayed in the same window. Statistics The Statistics function on the Audio menu analyzes the selected audio (events, clips, or range selections) and displays a window with the following information: Channel The name of the analyzed channel. Min. Sample Value The lowest sample value in dB. Max. Sample Value The highest sample value in dB. Peak Amplitude The largest amplitude in dB.
Audio processing and functions Statistics 427 True Peak The maximum absolute level of the audio signal waveform in the continuous time domain. DC Offset The amount of DC Offset as a percentage and in dB. Resolution The current calculated audio resolution. Estimated Pitch The estimated pitch. Sample Rate The sample rate. Average RMS (AES-17) The average loudness in accordance with the AES-17 standard. Max. RMS The highest RMS value. Max. RMS All Channels The highest RMS value of all channels. Integrated Loudness (Cubase Pro only) The average loudness over the whole title in LUFS (Loudness Unit, referenced to Full Scale) in accordance with EBU R-128 that recommends to normalize audio at -23 LUFS (±1 LU). Loudness Range (Cubase Pro only) The dynamic range over the whole title in LU (Loudness Units). This value allows you to see if dynamic processing is needed. Max. True Peak Level (Cubase Pro only) The maximum value of the audio signal waveform in the continuous time domain. Max. Momentary Loudness (Cubase Pro only) The maximum value of all momentary loudness values, based on a time window of 400 ms. The measurement is not gated. Max. Short-Term Loudness (Cubase Pro only) The maximum value of all short-term loudness values, based on a time window of 3 s. The measurement is not gated. RELATED LINKS Remove DC Offset on page 412
Audio processing and functions About time stretch and pitch shift algorithms 428 About time stretch and pitch shift algorithms In Cubase, time stretching and pitch shifting algorithms are used for numerous operations (e. g. the Time Stretch and Pitch Shift offline processes, in the Sample Editor, or by the Flatten function). Depending on the feature, some or all of the following algorithm presets are available. élastique The élastique algorithm is suited for both polyphonic and monophonic material. The algorithm has three modes, and there are three presets for each mode. The following modes are available: • élastique Pro – This mode offers the best audio quality, without formant preservation. • élastique Pro Formant – This is the same as the Pro mode, but including formant preservation. • élastique efficient – This mode requires less computing powers, but has a lower audio quality than the Pro modes. These modes are available with the following variants: • Time – Timing accuracy is favored over pitch accuracy. • Pitch – Pitch accuracy is favored over timing accuracy. • Tape – The pitch shift is locked to the time stretch (as when playing back a tape with varying speed). Stretching the audio material to a longer duration automatically decreases its pitch. This variant has no effect when used in combination with event transpose or the transpose track. MPEX MPEX is an alternative high-quality algorithm. You can choose between the following quality settings: MPEX – Preview Quality Use this mode only for preview purposes. MPEX – Mix Fast This mode is a very fast mode for preview. This works best with composite music signals (mono or stereo material). MPEX – Solo Fast Use this mode for single instruments (monophonic material) and voice.
Audio processing and functions About time stretch and pitch shift algorithms 429 MPEX – Solo Musical Same as above but higher quality. MPEX – Poly Fast Use this for processing monophonic and polyphonic material. This is the fastest setting that gives still very good results. You can use this for drum loops, mixes, chords. MPEX – Poly Musical Use this for processing monophonic and polyphonic material. This is the recommended MPEX default quality setting. You can use this for drum loops, mixes, chords. MPEX – Poly Complex This high quality setting is quite CPU-intensive and should be used only when processing difficult material or for stretch factors above 1.3. NOTE When applying the Pitch Shift process, you can choose between the regular setting and a setting where the formants are preserved for each quality setting. Standard The Standard algorithm is optimized for CPU efficient realtime processing. The following presets are available: Standard – Drums This mode is best for percussive sounds, because it does not change the timing of your audio. Using this option with certain tuned percussion instruments may lead to audible artifacts. In this case, try the Mix mode as an alternative. Standard – Plucked Use this mode for audio with transients and a relatively stable spectral sound character (e. g. plucked instruments). Standard – Pads Use this mode for pitched audio with slower rhythm and a stable spectral sound character. This minimizes sound artifacts, but the rhythmic accuracy is not preserved. Standard – Vocals This mode is suitable for slower signals with transients and a prominent tonal character (e. g. vocals).
Audio processing and functions About time stretch and pitch shift algorithms 430 Standard – Mix This mode preserves the rhythm and minimizes the artifacts for pitched material that does not meet the above criteria (i. e. with a less homogenous sound character). This preset is selected by default for audio that is not categorized. Standard – Custom This preset allows you to manually tweak the time stretching parameters (see below). By default, the settings that are shown when you open the dialog are those of the last preset used (except if the Solo preset has been selected, see below). Standard – Solo This mode preserves the timbre of the audio. Only use it for monophonic material (solo woodwind/brass instruments or solo vocals, monophonic synths or string instruments that do not play harmonies). If you select the “Standard – Custom” option, a dialog opens where you can manually adjust the three parameters that govern the sound quality of the time stretching: Grain size The standard time-stretching algorithm splits the audio into small pieces called “grains”. This parameter determines the size of the grains. For material with many transients, use low grain size values for best results. Overlap Overlap is the percentage of the whole grain that will overlap with other grains. Use higher values for material with a stable sound character. Variance Variance is also a percentage of the whole length of the grains, and sets a variation in positioning so that the overlapping area sounds smooth. A Variance setting of 0 will produce a sound akin to time stretching used in early samplers, whereas higher settings produce more (rhythmic) “smearing” effects but less audio artifacts. Limitations Applying time stretching or pitch shifting to audio material can lead to a degradation in audio quality and to audible artifacts. The result depends on many factors, such as the source material, the particular stretch and pitch operations applied, and the selected audio algorithm preset. As a rule of thumb, smaller changes in pitch or duration cause less degradation. However, there are additional issues one should be aware of when working with time stretching and pitch shifting algorithms.