Generating an inverted phase of a sound wave involves shifting the phase of the wave by 180 degrees. Here's a general method to achieve this:
Obtain the audio signal: Start by obtaining the digital representation of the sound wave you want to invert. This can be done by either recording the sound using a microphone or loading an existing audio file into a digital audio workstation (DAW) or programming environment.
Convert the audio to a numerical representation: Convert the audio signal into a numerical representation, such as a series of samples. Each sample represents the amplitude of the sound wave at a specific point in time.
Apply phase inversion: To invert the phase of the sound wave, you need to multiply each sample by -1. This will reverse the polarity of the waveform, effectively inverting the phase. In mathematical terms, the formula for phase inversion is:
inverted_sample = -1 * original_sample
Apply this formula to each sample in the audio signal.
Optional: Handling audio normalization: In some cases, the inverted waveform might have a different amplitude range compared to the original waveform. To ensure both waveforms have similar loudness, you can apply audio normalization techniques, such as peak normalization or RMS normalization. These techniques adjust the amplitude of the inverted waveform to match the desired level.
Optional: Export the inverted audio: If you're working with a DAW or programming environment that allows audio export, you can save the inverted audio waveform as a new audio file. This will enable you to use it for further analysis, processing, or playback.
Remember that the phase inversion process described above will invert the entire waveform, resulting in a complete reversal of the sound's phase. If you want to invert the phase of only a specific frequency range or time segment, additional signal processing techniques like filtering or time-domain operations may be required.