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//! Double precision / f64 / double. /// Applies a double precision (double, f64) low pass filter on a vector of **mono sample** in 16 /// bit resolution. If you have stereo data, call this function for each channel, convert it first /// to mono or do whatever fits your use case. /// /// ## Parameters /// * `data` audio samples /// * `sample_rate_hz` Sample Rate, e.g. 44100Hz /// * `cutoff_frequency_hz` upper bound for frequencies to be cut, e.g. 150Hz pub fn apply_lpf_i16_dp(data: &mut [i16], sample_rate_hz: u16, cutoff_frequency_hz: u16) { // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter let rc = 1.0 / (cutoff_frequency_hz as f64 * 2.0 * core::f64::consts::PI); let dt = 1.0 / sample_rate_hz as f64; let alpha = dt / (rc + dt); data[0] = (alpha * data[0] as f64) as i16; for i in 1..data.len() { // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter // we don't need a copy of the original data, because the original data is accessed // before it is overwritten: data[i] = ... data[i] data[i] = (data[i - 1] as f64 + alpha * (data[i] as f64 - data[i - 1] as f64)) as i16; } } /// Same as [`apply_lpf_i16_dp`] but with i32 audio resolution. pub fn apply_lpf_i32_dp(data: &mut [i32], sample_rate_hz: u16, cutoff_frequency_hz: u16) { // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter let rc = 1.0 / (cutoff_frequency_hz as f64 * 2.0 * core::f64::consts::PI); let dt = 1.0 / sample_rate_hz as f64; let alpha = dt / (rc + dt); data[0] = (alpha * data[0] as f64) as i32; for i in 1..data.len() { // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter // we don't need a copy of the original data, because the original data is accessed // before it is overwritten: data[i] = ... data[i] data[i] = (data[i - 1] as f64 + alpha * (data[i] as f64 - data[i - 1] as f64)) as i32; } }