Adobe Audition provides several ways to analyze audio. To compare phase relationships between any two channels, use the Phase Meter panel. To analyze tonal and dynamic range, use the Frequency Analysis and Amplitude Statistics panels.
The Waveform Editor also provides Spectral Frequency Display, which you can use together with the analysis methods above. (See Displaying audio in the Waveform Editor.)
The Phase Meter panel reveals out‑of‑phase channels for stereo and surround waveforms, which you can address with the Effects > Invert command. (See Invert a waveform.) This panel also helps you identify highly in-phase channels that will sound similar if summed to mono. (See Convert a waveform between surround, stereo, and mono.)
To understand audio phase, see How sound waves interact.
In the Phase Meter, audio to the left is more out of phase, while audio to the right is more in phase. -1.0 reflects total phase cancellation, while 1.0 reflects identical audio content in each channel.
To customize meter appearance, right-click them, and select Show Color Gradient or Show LED Meters.
You can use the Frequency Analysis panel to identify problematic frequency bands, which you can then correct with a filter effect.
Displays the frequency scale either logarithmically (reflecting human hearing) or linearly (providing more detail for upper frequencies).
Let you take up to eight frequency snapshots as a waveform is playing. The frequency outline (which is rendered in the same color as the button clicked) is frozen on the graph and overlaid on other frequency outlines. To clear a frozen frequency outline, click its corresponding Hold button again.
Determines which channel of a stereo or surround file appears over others in the graph. To combine displayed channels, choose Average.
Scan or Scan Selection
Scans the entire file or selection, and displays average frequency data in the graph. (By default, the graph displays data from the center point of files and selections.)
Specifies the Fast Fourier Transform size. Higher FFT sizes report frequency data more accurately but they require longer processing times.
Determines the Fast Fourier transform shape. These functions are listed in order from narrowest to widest. Narrower functions include fewer surrounding frequencies but less precisely reflect center frequencies. Wider functions include more surrounding frequencies but more precisely reflect center frequencies. The Hamming and Blackman options provide excellent overall results.
0 dB Reference
Determines the amplitude at which full scale, 0 dBFS audio data is displayed. For example, a value of zero displays 0 dBFS audio at 0 dB. A value of 30 displays 0 dBFS audio at –30 dB. This value simply moves the graph up or down; it does not change the amplitude of audio data.
Tip: Adjust the 0 dB Reference to calibrate this display to another decibel reference, like sound pressure level (SPL).
Value at [x] Hz
Reveals precise amplitude for specific frequencies when you position the mouse over the graph.
Overall Musical Note
For the start point of a selected range, indicates keyboard position and variance from standard tuning (A440). For example, A2 +7 equals the second‑lowest A on a keyboard tuned 7% higher than normal.
To zoom in on a graph, right‑click and drag the magnifying glass icon in the vertical or horizontal ruler.
To navigate a magnified graph, left‑click and drag the hand icon in the vertical or horizontal ruler.
To zoom out on a magnified graph, right‑click in the vertical or horizontal ruler, and choose Zoom Out to return to the previous magnification, or Zoom Out Full to zoom out completely.
The General tab displays numerical statistics that indicate dynamic range, identify clipped samples, and note any DC offset.
The RMS Histogram tab displays a graph that shows the relative prevalence of each amplitude. The horizontal ruler measures amplitude in decibels, and the vertical ruler measures prevalence using the RMS formula. Choose a channel to display from the Show Channel menu.
Tip: Use the Histogram tab to identify prevalent amplitudes, and then compress, limit, or normalize them with an amplitude effect.
Possibly Clipped Samples
Shows the number of samples have likely exceeded 0 dBFS. Click the icon to the right of this value to navigate to the first clipped sample in the audio file. (If necessary, click the icon again to view subsequent clipped samples.)
Total, Maximum, Minimum, and Average RMS Amplitude
Show the root-mean-square values of the selection. RMS values are based on the prevalence of specific amplitudes, often reflecting perceived loudness better than absolute or average amplitudes.
Shows any direct current offset applied to the waveform during recording. Positive values are above the center line, and negative values are below it. (See Correct DC offset.)
Measured Bit Depth
Reports the waveform’s bit depth. (32 indicates that the waveform uses the full 32‑bit float range).
Dynamic Range Used
Shows the dynamic range minus unusually long periods of low RMS amplitude, such as silent passages.
0dB = FS Sine Wave
Correspond the dB level to a full‑scale sine wave, where peak amplitude is about 3.01 dB quieter than a full-scale square wave.
0dB = FS Square Wave
Corresponds the dB level to a full‑scale square wave, where peak amplitude is about 3.01 dB louder than a full‑scale sine wave.
Specifies the number of milliseconds in each RMS window. A selected range contains a series of such windows, which Adobe Audition averages to calculate the Minimum RMS and Maximum RMS values. To achieve the most accurate RMS values, use wide windows for audio with a wide dynamic range, and narrow windows for audio with a narrow dynamic range.