diff --git a/audiomentations/augmentations/add_color_noise.py b/audiomentations/augmentations/add_color_noise.py
index 2facc249..1cd51c01 100644
--- a/audiomentations/augmentations/add_color_noise.py
+++ b/audiomentations/augmentations/add_color_noise.py
@@ -174,7 +174,7 @@ def __init__(
     def randomize_parameters(self, samples: np.ndarray, sample_rate: int):
         super().randomize_parameters(samples, sample_rate)
         if self.parameters["should_apply"]:
-            # Pick SNR in decibel scale
+            # Pick SNR in Decibel scale
             snr = random.uniform(self.min_snr_db, self.max_snr_db)
 
             # Pick f_decay
diff --git a/audiomentations/augmentations/add_gaussian_snr.py b/audiomentations/augmentations/add_gaussian_snr.py
index 8806f0e4..73e96f6e 100644
--- a/audiomentations/augmentations/add_gaussian_snr.py
+++ b/audiomentations/augmentations/add_gaussian_snr.py
@@ -10,7 +10,7 @@
 class AddGaussianSNR(BaseWaveformTransform):
     """
     Add gaussian noise to the input. A random Signal to Noise Ratio (SNR) will be picked
-    uniformly in the decibel scale. This aligns with human hearing, which is more
+    uniformly in the Decibel scale. This aligns with human hearing, which is more
     logarithmic than linear.
     """
 
diff --git a/audiomentations/augmentations/apply_impulse_response.py b/audiomentations/augmentations/apply_impulse_response.py
index 19de1e0a..122b1d5b 100644
--- a/audiomentations/augmentations/apply_impulse_response.py
+++ b/audiomentations/augmentations/apply_impulse_response.py
@@ -46,8 +46,6 @@ def __init__(
         self.__load_ir = functools.lru_cache(maxsize=lru_cache_size)(self.__load_ir)
         self.leave_length_unchanged = leave_length_unchanged
 
-        self.leave_length_unchanged = leave_length_unchanged
-
     @staticmethod
     def __load_ir(file_path, sample_rate, mono):
         return load_sound_file(file_path, sample_rate, mono=mono)
@@ -58,7 +56,8 @@ def randomize_parameters(self, samples: NDArray[np.float32], sample_rate: int):
             self.parameters["ir_file_path"] = random.choice(self.ir_files)
 
     def apply(self, samples: NDArray[np.float32], sample_rate: int):
-        load_mono_ir = samples.ndim == 1 # determine if ir should load as mono
+        # Determine if the impulse response should be loaded as mono
+        load_mono_ir = samples.ndim == 1
         ir, sample_rate2 = self.__load_ir(self.parameters["ir_file_path"], sample_rate, mono=load_mono_ir)
         if sample_rate != sample_rate2:
             # This will typically not happen, as librosa should automatically resample the
@@ -72,7 +71,7 @@ def apply(self, samples: NDArray[np.float32], sample_rate: int):
         samples_original_dim = samples.ndim
         samples, ir = np.atleast_2d(samples), np.atleast_2d(ir)
 
-        # Preallocate the the output array
+        # Preallocate the output array
         output_shape = (samples.shape[0], samples.shape[1] + ir.shape[1] - 1)
         signal_ir = np.empty(output_shape, dtype=samples.dtype)
 
diff --git a/audiomentations/augmentations/band_pass_filter.py b/audiomentations/augmentations/band_pass_filter.py
index b48d20ec..d749984a 100644
--- a/audiomentations/augmentations/band_pass_filter.py
+++ b/audiomentations/augmentations/band_pass_filter.py
@@ -31,10 +31,10 @@ def __init__(
         :param max_rolloff: Maximum filter roll-off (in dB/octave)
             Must be a multiple of 6
         :param zero_phase: Whether filtering should be zero phase.
-            When this is set to `True` it will not affect the phase of the
+            When this is set to `True`, it will not affect the phase of the
             input signal but will sound 3 dB lower at the cutoff frequency
             compared to the non-zero phase case (6 dB vs 3 dB). Additionally,
-            it is 2 times slower than in the non-zero phase case. If you
+            it is twice as slow as the non-zero phase case. If you
             absolutely want no phase distortions (e.g. want to augment an
             audio file with lots of transients, like a drum track), set
             this to `True`.
diff --git a/audiomentations/augmentations/band_stop_filter.py b/audiomentations/augmentations/band_stop_filter.py
index 67a6c1f1..b4e83b4a 100644
--- a/audiomentations/augmentations/band_stop_filter.py
+++ b/audiomentations/augmentations/band_stop_filter.py
@@ -36,12 +36,12 @@ def __init__(
         :param max_rolloff: Maximum filter roll-off (in dB/octave)
             Must be a multiple of 6
         :param zero_phase: Whether filtering should be zero phase.
-            When this is set to `true` it will not affect the phase of the
+            When this is set to `True`, it will not affect the phase of the
             input signal but will sound 3 dB lower at the cutoff frequency
             compared to the non-zero phase case (6 dB vs 3 dB). Additionally,
-            it is 2 times slower than in the non-zero phase case. If you
+            it is twice as slow as the non-zero phase case. If you
             absolutely want no phase distortions (e.g. want to augment a
-            drum track), set this to `true`.
+            drum track), set this to `True`.
         :param p: The probability of applying this transform
         """
         super().__init__(
diff --git a/audiomentations/augmentations/base_butterword_filter.py b/audiomentations/augmentations/base_butterword_filter.py
index e4724526..f9ad0cbb 100644
--- a/audiomentations/augmentations/base_butterword_filter.py
+++ b/audiomentations/augmentations/base_butterword_filter.py
@@ -188,7 +188,7 @@ def apply(self, samples: NDArray[np.float32], sample_rate: int = None):
             cutoff_freq = self.parameters["cutoff_freq"]
             nyquist_freq = sample_rate // 2
             if cutoff_freq > nyquist_freq:
-                # Ensure that the cutoff frequency does not exceed the nyquist
+                # Ensure that the cutoff frequency does not exceed the Nyquist
                 # frequency to avoid an exception from scipy
                 cutoff_freq = nyquist_freq * 0.9999
             sos = butter(
@@ -206,7 +206,7 @@ def apply(self, samples: NDArray[np.float32], sample_rate: int = None):
             )
             nyquist_freq = sample_rate // 2
             if high_freq > nyquist_freq:
-                # Ensure that the upper critical frequency does not exceed the nyquist
+                # Ensure that the upper critical frequency does not exceed the Nyquist
                 # frequency to avoid an exception from scipy
                 high_freq = nyquist_freq * 0.9999
             sos = butter(
diff --git a/audiomentations/augmentations/high_pass_filter.py b/audiomentations/augmentations/high_pass_filter.py
index 8e0d6709..1ba7520b 100644
--- a/audiomentations/augmentations/high_pass_filter.py
+++ b/audiomentations/augmentations/high_pass_filter.py
@@ -26,12 +26,12 @@ def __init__(
         :param max_rolloff: Maximum filter roll-off (in dB/octave)
             Must be a multiple of 6
         :param zero_phase: Whether filtering should be zero phase.
-            When this is set to `true` it will not affect the phase of the
+            When this is set to `True`, it will not affect the phase of the
             input signal but will sound 3 dB lower at the cutoff frequency
             compared to the non-zero phase case (6 dB vs. 3 dB). Additionally,
-            it is 2 times slower than in the non-zero phase case. If you
+            it is twice as slow as the non-zero phase case. If you
             absolutely want no phase distortions (e.g. want to augment a
-            drum track), set this to `true`.
+            drum track), set this to `True`.
         :param p: The probability of applying this transform
         """
         super().__init__(
diff --git a/audiomentations/augmentations/high_shelf_filter.py b/audiomentations/augmentations/high_shelf_filter.py
index 58ae07c0..6461c95c 100644
--- a/audiomentations/augmentations/high_shelf_filter.py
+++ b/audiomentations/augmentations/high_shelf_filter.py
@@ -16,7 +16,7 @@ class HighShelfFilter(BaseWaveformTransform):
     A high shelf filter is a filter that either boosts (increases amplitude) or cuts
     (decreases amplitude) frequencies above a certain center frequency. This transform
     applies a high-shelf filter at a specific center frequency in hertz.
-    The gain at nyquist frequency is controlled by `{min,max}_gain_db` (note: can be positive or negative!).
+    The gain at Nyquist frequency is controlled by `{min,max}_gain_db` (note: can be positive or negative!).
     Filter coefficients are taken from the W3 Audio EQ Cookbook: https://www.w3.org/TR/audio-eq-cookbook/
     """
 
@@ -35,8 +35,8 @@ def __init__(
         """
         :param min_center_freq: The minimum center frequency of the shelving filter
         :param max_center_freq: The maximum center frequency of the shelving filter
-        :param min_gain_db: The minimum gain at the nyquist frequency
-        :param max_gain_db: The maximum gain at the nyquist frequency
+        :param min_gain_db: The minimum gain at the Nyquist frequency
+        :param max_gain_db: The maximum gain at the Nyquist frequency
         :param min_q: The minimum quality factor Q. The higher the Q, the steeper the
             transition band will be.
         :param max_q: The maximum quality factor Q. The higher the Q, the steeper the
@@ -120,7 +120,7 @@ def apply(self, samples: NDArray[np.float32], sample_rate: int):
         nyquist_freq = sample_rate // 2
         center_freq = self.parameters["center_freq"]
         if center_freq > nyquist_freq:
-            # Ensure that the center frequency is below the nyquist
+            # Ensure that the center frequency is below the Nyquist
             # frequency to avoid filter instability
             center_freq = nyquist_freq * 0.9999
 
diff --git a/audiomentations/augmentations/limiter.py b/audiomentations/augmentations/limiter.py
index a1722a59..b86749d5 100644
--- a/audiomentations/augmentations/limiter.py
+++ b/audiomentations/augmentations/limiter.py
@@ -15,15 +15,14 @@
 class Limiter(BaseWaveformTransform):
     """
     A simple audio limiter (dynamic range compression).
-    Note: This transform also delays the signal by a fraction of the attack time.
     """
 
     supports_multichannel = True
 
     def __init__(
         self,
-        min_threshold_db: float = -24,
-        max_threshold_db: float = -2,
+        min_threshold_db: float = -24.0,
+        max_threshold_db: float = -2.0,
         min_attack: float = 0.0005,
         max_attack: float = 0.025,
         min_release: float = 0.05,
@@ -36,8 +35,8 @@ def __init__(
         The attack time is how quickly the limiter kicks in once the audio signal starts exceeding the threshold.
         The release time determines how quickly the limiter stops working after the signal drops below the threshold.
 
-        :param min_threshold_db: Minimum threshold in decibels
-        :param max_threshold_db: Maximum threshold in decibels
+        :param min_threshold_db: Minimum threshold in Decibels
+        :param max_threshold_db: Maximum threshold in Decibels
         :param min_attack: Minimum attack time in seconds
         :param max_attack: Maximum attack time in seconds
         :param min_release: Minimum release time in seconds
diff --git a/audiomentations/augmentations/low_pass_filter.py b/audiomentations/augmentations/low_pass_filter.py
index 9000100d..0a9303ba 100644
--- a/audiomentations/augmentations/low_pass_filter.py
+++ b/audiomentations/augmentations/low_pass_filter.py
@@ -26,12 +26,12 @@ def __init__(
         :param max_rolloff: Maximum filter roll-off (in dB/octave)
             Must be a multiple of 6
         :param zero_phase: Whether filtering should be zero phase.
-            When this is set to `true` it will not affect the phase of the
+            When this is set to `True`, it will not affect the phase of the
             input signal but will sound 3 dB lower at the cutoff frequency
             compared to the non-zero phase case (6 dB vs. 3 dB). Additionally,
-            it is 2 times slower than in the non-zero phase case. If you
+            it is twice as slow as the non-zero phase case. If you
             absolutely want no phase distortions (e.g. want to augment a
-            drum track), set this to `true`.
+            drum track), set this to `True`.
         :param p: The probability of applying this transform
         """
         super().__init__(
diff --git a/audiomentations/augmentations/low_shelf_filter.py b/audiomentations/augmentations/low_shelf_filter.py
index 7ba0318f..bb2cd8bb 100644
--- a/audiomentations/augmentations/low_shelf_filter.py
+++ b/audiomentations/augmentations/low_shelf_filter.py
@@ -119,7 +119,7 @@ def apply(self, samples: NDArray[np.float32], sample_rate: int):
         nyquist_freq = sample_rate // 2
         center_freq = self.parameters["center_freq"]
         if center_freq > nyquist_freq:
-            # Ensure that the center frequency is below the nyquist
+            # Ensure that the center frequency is below the Nyquist
             # frequency to avoid filter instability
             center_freq = nyquist_freq * 0.9999
 
diff --git a/audiomentations/augmentations/trim.py b/audiomentations/augmentations/trim.py
index c379b921..32a496eb 100644
--- a/audiomentations/augmentations/trim.py
+++ b/audiomentations/augmentations/trim.py
@@ -14,7 +14,7 @@ class Trim(BaseWaveformTransform):
 
     def __init__(self, top_db: float = 30.0, p: float = 0.5):
         """
-        :param top_db: The threshold (in decibels) below reference to consider as silence
+        :param top_db: The threshold (in Decibels) below reference to consider as silence
         :param p: The probability of applying this transform
         """
         super().__init__(p)
diff --git a/docs/changelog.md b/docs/changelog.md
index 47fa3ec6..3e9a546d 100644
--- a/docs/changelog.md
+++ b/docs/changelog.md
@@ -323,7 +323,7 @@ These are **breaking changes**. The following example shows how you can adapt yo
 * When looking for audio files in `AddImpulseResponse`, `AddBackgroundNoise`
   and `AddShortNoises`, follow symlinks by default.
 * When using the new parameters `min_snr_in_db` and `max_snr_in_db` in `AddGaussianSNR`,
-  SNRs will be picked uniformly in _the decibel scale_ instead of in the linear amplitude
+  SNRs will be picked uniformly in _the Decibel scale_ instead of in the linear amplitude
   ratio scale. The new behavior aligns more with human hearing, which is not linear.
 
 ### Fixed
diff --git a/docs/waveform_transforms/add_background_noise.md b/docs/waveform_transforms/add_background_noise.md
index de4754c5..0aefeaea 100644
--- a/docs/waveform_transforms/add_background_noise.md
+++ b/docs/waveform_transforms/add_background_noise.md
@@ -26,7 +26,7 @@ Here are some examples of datasets that can be downloaded and used as background
 ## Input-output example
 
 Here we add some music to a speech recording, targeting a signal-to-noise ratio (SNR) of
-5 decibels (dB), which means that the speech (_signal_) is 5 dB louder than the music (_noise_).
+5 Decibels (dB), which means that the speech (_signal_) is 5 dB louder than the music (_noise_).
 
 ![Input-output waveforms and spectrograms](AddBackgroundNoise.webp)
 
diff --git a/docs/waveform_transforms/add_gaussian_snr.md b/docs/waveform_transforms/add_gaussian_snr.md
index 7eba8cca..f02c210d 100644
--- a/docs/waveform_transforms/add_gaussian_snr.md
+++ b/docs/waveform_transforms/add_gaussian_snr.md
@@ -4,7 +4,7 @@ _Added in v0.7.0_
 
 The `AddGaussianSNR` transform injects Gaussian noise into an audio signal. It applies
 a **Signal-to-Noise Ratio (SNR)** that is chosen randomly from a **uniform distribution on the
-decibel scale**. This choice is consistent with the nature of human hearing, which is
+Decibel scale**. This choice is consistent with the nature of human hearing, which is
 logarithmic rather than linear.
 
 **SNR** is a common measure used in science and engineering to compare the level of a
diff --git a/docs/waveform_transforms/apply_impulse_response.md b/docs/waveform_transforms/apply_impulse_response.md
index 917ae4b4..e8a48a2c 100644
--- a/docs/waveform_transforms/apply_impulse_response.md
+++ b/docs/waveform_transforms/apply_impulse_response.md
@@ -46,7 +46,7 @@ will slow down execution, and because some high frequencies may get lost.
 
 ## Input-output example
 
-Here we make a dry speech recording quite reverbant by convolving it with a room impulse response
+Here we make a dry speech recording quite reverberant by convolving it with a room impulse response
 
 ![Input-output waveforms and spectrograms](ApplyImpulseResponse.webp)
 
diff --git a/docs/waveform_transforms/band_pass_filter.md b/docs/waveform_transforms/band_pass_filter.md
index 2adacd44..3b8cc619 100644
--- a/docs/waveform_transforms/band_pass_filter.md
+++ b/docs/waveform_transforms/band_pass_filter.md
@@ -46,19 +46,19 @@ augmented_sound = transform(my_waveform_ndarray, sample_rate=48000)
 [`max_bandwidth_fraction`](#max_bandwidth_fraction){ #max_bandwidth_fraction }: `float` • range: [0.0, 2.0]
 :   :octicons-milestone-24: Default: `1.99`. Maximum bandwidth relative to center frequency
 
-[`min_rolloff`](#min_rolloff){ #min_rolloff }: `float` • unit: Decibels/octave
+[`min_rolloff`](#min_rolloff){ #min_rolloff }: `int` • unit: Decibels/octave
 :   :octicons-milestone-24: Default: `12`. Minimum filter roll-off (in dB/octave).
-    Must be a multiple of 6
+    Must be a multiple of 6 (or 12 if `zero_phase` is `True`)
 
-[`max_rolloff`](#max_rolloff){ #max_rolloff }: `float` • unit: Decibels/octave
+[`max_rolloff`](#max_rolloff){ #max_rolloff }: `int` • unit: Decibels/octave
 :   :octicons-milestone-24: Default: `24`. Maximum filter roll-off (in dB/octave)
-    Must be a multiple of 6
+    Must be a multiple of 6 (or 12 if `zero_phase` is `True`)
 
 [`zero_phase`](#zero_phase){ #zero_phase }: `bool`
 :   :octicons-milestone-24: Default: `False`. Whether filtering should be zero phase.
-    When this is set to `True` it will not affect the phase of the input signal but will
+    When this is set to `True`, it will not affect the phase of the input signal but will
     sound 3 dB lower at the cutoff frequency compared to the non-zero phase case (6 dB
-    vs. 3 dB). Additionally, it is 2 times slower than in the non-zero phase case. If
+    vs. 3 dB). Additionally, it is twice as slow as the non-zero phase case. If
     you absolutely want no phase distortions (e.g. want to augment an audio file with
     lots of transients, like a drum track), set this to `True`.
 
diff --git a/docs/waveform_transforms/band_stop_filter.md b/docs/waveform_transforms/band_stop_filter.md
index 1e9fbf3b..ce83d5f2 100644
--- a/docs/waveform_transforms/band_stop_filter.md
+++ b/docs/waveform_transforms/band_stop_filter.md
@@ -36,24 +36,24 @@ can hear that the timbre is different in the transformed sound than in the origi
 :   :octicons-milestone-24: Default: `4000.0`. Maximum center frequency in hertz
 
 [`min_bandwidth_fraction`](#min_bandwidth_fraction){ #min_bandwidth_fraction }: `float`
-:   :octicons-milestone-24: Default: `0.5`. Minimum bandwidth relative to center frequency
+:   :octicons-milestone-24: Default: `0.5`. Minimum bandwidth fraction relative to center frequency
 
 [`max_bandwidth_fraction`](#max_bandwidth_fraction){ #max_bandwidth_fraction }: `float`
-:   :octicons-milestone-24: Default: `1.99`. Maximum bandwidth relative to center frequency
+:   :octicons-milestone-24: Default: `1.99`. Maximum bandwidth fraction relative to center frequency
 
-[`min_rolloff`](#min_rolloff){ #min_rolloff }: `float` • unit: Decibels/octave
+[`min_rolloff`](#min_rolloff){ #min_rolloff }: `int` • unit: Decibels/octave
 :   :octicons-milestone-24: Default: `12`. Minimum filter roll-off (in dB/octave).
-    Must be a multiple of 6
+    Must be a multiple of 6 (or 12 if `zero_phase` is `True`)
 
-[`max_rolloff`](#max_rolloff){ #max_rolloff }: `float` • unit: Decibels/octave
+[`max_rolloff`](#max_rolloff){ #max_rolloff }: `int` • unit: Decibels/octave
 :   :octicons-milestone-24: Default: `24`. Maximum filter roll-off (in dB/octave)
-    Must be a multiple of 6
+    Must be a multiple of 6 (or 12 if `zero_phase` is `True`)
 
 [`zero_phase`](#zero_phase){ #zero_phase }: `bool`
 :   :octicons-milestone-24: Default: `False`. Whether filtering should be zero phase.
-    When this is set to `True` it will not affect the phase of the input signal but will
+    When this is set to `True`, it will not affect the phase of the input signal but will
     sound 3 dB lower at the cutoff frequency compared to the non-zero phase case (6 dB
-    vs. 3 dB). Additionally, it is 2 times slower than in the non-zero phase case. If
+    vs. 3 dB). Additionally, it is twice as slow as the non-zero phase case. If
     you absolutely want no phase distortions (e.g. want to augment an audio file with
     lots of transients, like a drum track), set this to `True`.
 
diff --git a/docs/waveform_transforms/high_pass_filter.md b/docs/waveform_transforms/high_pass_filter.md
index 2a7a853b..cb2908f2 100644
--- a/docs/waveform_transforms/high_pass_filter.md
+++ b/docs/waveform_transforms/high_pass_filter.md
@@ -15,17 +15,17 @@ Can also be set for zero-phase filtering (will result in a 6 dB drop at cutoff).
 
 [`min_rolloff`](#min_rolloff){ #min_rolloff }: `float` • unit: Decibels/octave
 :   :octicons-milestone-24: Default: `12`. Minimum filter roll-off (in dB/octave).
-    Must be a multiple of 6
+    Must be a multiple of 6 (or 12 if `zero_phase` is `True`)
 
 [`max_rolloff`](#max_rolloff){ #max_rolloff }: `float` • unit: Decibels/octave
 :   :octicons-milestone-24: Default: `24`. Maximum filter roll-off (in dB/octave).
-    Must be a multiple of 6
+    Must be a multiple of 6 (or 12 if `zero_phase` is `True`)
 
 [`zero_phase`](#zero_phase){ #zero_phase }: `bool`
 :   :octicons-milestone-24: Default: `False`. Whether filtering should be zero phase.
-    When this is set to `True` it will not affect the phase of the input signal but will
+    When this is set to `True`, it will not affect the phase of the input signal but will
     sound 3 dB lower at the cutoff frequency compared to the non-zero phase case (6 dB
-    vs. 3 dB). Additionally, it is 2 times slower than in the non-zero phase case. If
+    vs. 3 dB). Additionally, it is twice as slow as the non-zero phase case. If
     you absolutely want no phase distortions (e.g. want to augment an audio file with
     lots of transients, like a drum track), set this to `True`.
 
diff --git a/docs/waveform_transforms/low_pass_filter.md b/docs/waveform_transforms/low_pass_filter.md
index ca87ab66..c6d6db1c 100644
--- a/docs/waveform_transforms/low_pass_filter.md
+++ b/docs/waveform_transforms/low_pass_filter.md
@@ -13,19 +13,19 @@ Can also be set for zero-phase filtering (will result in a 6 dB drop at cutoff).
 [`max_cutoff_freq`](#max_cutoff_freq){ #max_cutoff_freq }: `float` • unit: hertz
 :   :octicons-milestone-24: Default: `7500.0`. Maximum cutoff frequency
 
-[`min_rolloff`](#min_rolloff){ #min_rolloff }: `float` • unit: Decibels/octave
+[`min_rolloff`](#min_rolloff){ #min_rolloff }: `int` • unit: Decibels/octave
 :   :octicons-milestone-24: Default: `12`. Minimum filter roll-off (in dB/octave).
-    Must be a multiple of 6
+    Must be a multiple of 6 (or 12 if `zero_phase` is `True`)
 
-[`max_rolloff`](#max_rolloff){ #max_rolloff }: `float` • unit: Decibels/octave
+[`max_rolloff`](#max_rolloff){ #max_rolloff }: `int` • unit: Decibels/octave
 :   :octicons-milestone-24: Default: `24`. Maximum filter roll-off (in dB/octave)
-    Must be a multiple of 6
+    Must be a multiple of 6 (or 12 if `zero_phase` is `True`)
 
 [`zero_phase`](#zero_phase){ #zero_phase }: `bool`
 :   :octicons-milestone-24: Default: `False`. Whether filtering should be zero phase.
-    When this is set to `True` it will not affect the phase of the input signal but will
+    When this is set to `True`, it will not affect the phase of the input signal but will
     sound 3 dB lower at the cutoff frequency compared to the non-zero phase case (6 dB
-    vs. 3 dB). Additionally, it is 2 times slower than in the non-zero phase case. If
+    vs. 3 dB). Additionally, it is twice as slow as the non-zero phase case. If
     you absolutely want no phase distortions (e.g. want to augment an audio file with
     lots of transients, like a drum track), set this to `True`.
 
diff --git a/docs/waveform_transforms/normalize.md b/docs/waveform_transforms/normalize.md
index 57c703a8..f6d47542 100644
--- a/docs/waveform_transforms/normalize.md
+++ b/docs/waveform_transforms/normalize.md
@@ -8,6 +8,12 @@ Also known as peak normalization.
 
 # Normalize API
 
+[`apply_to`](#apply_to){ #apply_to }: `str` • choices: `"all"`, `"only_too_loud_sounds"`
+:   :octicons-milestone-24: Default: `"all"`. Defines the criterion for applying the transform.
+
+    * `"all"`: Apply peak normalization to all inputs
+    * `"only_too_loud_sounds"`: Apply peak normalization only to inputs where the maximum absolute peak is greater than 1
+
 [`p`](#p){ #p }: `float` • range: [0.0, 1.0]
 :   :octicons-milestone-24: Default: `0.5`. The probability of applying this transform.
 
diff --git a/docs/waveform_transforms/trim.md b/docs/waveform_transforms/trim.md
index a043e144..dbabe53a 100644
--- a/docs/waveform_transforms/trim.md
+++ b/docs/waveform_transforms/trim.md
@@ -3,7 +3,7 @@
 _Added in v0.7.0_
 
 Trim leading and trailing silence from an audio signal using `librosa.effects.trim`. It considers threshold
-(in decibels) below reference defined in parameter `top_db` as silence.
+(in Decibels) below reference defined in parameter `top_db` as silence.
 
 ## Input-output example
 
@@ -31,7 +31,7 @@ augmented_sound = transform(my_waveform_ndarray, sample_rate=16000)
 ## Trim API
 
 [`top_db`](#top_db){ #top_db }: `float` • unit: Decibel
-:   :octicons-milestone-24: Default: `30.0`. The threshold value (in decibels) below which to consider silence and trim.
+:   :octicons-milestone-24: Default: `30.0`. The threshold value (in Decibels) below which to consider silence and trim.
 
 [`p`](#p){ #p }: `float` • range: [0.0, 1.0]
 :   :octicons-milestone-24: Default: `0.5`. The probability of applying this transform.
diff --git a/tests/test_air_absorption.py b/tests/test_air_absorption.py
index f071f5e1..46157c63 100644
--- a/tests/test_air_absorption.py
+++ b/tests/test_air_absorption.py
@@ -10,7 +10,7 @@
 
 
 def get_chirp_test(sample_rate, duration):
-    """Create a `duration` seconds chirp from 0Hz to `nyquist frequency`"""
+    """Create a `duration` seconds chirp from 0Hz to `Nyquist frequency`"""
     n = np.arange(0, duration, 1 / sample_rate)
     samples = scipy.signal.chirp(n, 0, duration, sample_rate // 2, method="linear")
     return samples.astype(np.float32)
diff --git a/tests/test_filter_transforms.py b/tests/test_filter_transforms.py
index d8d05f51..2aa4c272 100644
--- a/tests/test_filter_transforms.py
+++ b/tests/test_filter_transforms.py
@@ -17,7 +17,7 @@
 
 
 def get_chirp_test(sample_rate, duration):
-    """Create a `duration` seconds chirp from 0Hz to `nyquist frequency`"""
+    """Create a `duration` seconds chirp from 0Hz to `Nyquist frequency`"""
     n = np.arange(0, duration, 1 / sample_rate)
     samples = scipy.signal.chirp(n, 0, duration, sample_rate // 2, method="linear")
     return samples.astype(np.float32)
@@ -738,7 +738,7 @@ def test_n_channel_input(self, samples, rolloff, zero_phase, num_channels):
     @pytest.mark.parametrize("zero_phase", [False])
     def test_nyquist_limit(self, cutoff_frequency, rolloff, zero_phase):
         # Test that the filter doesn't raise an exception when
-        # cutoff_frequency is greater than the nyquist frequency
+        # cutoff_frequency is greater than the Nyquist frequency
 
         sample_rate = 8000
 
@@ -947,7 +947,7 @@ def test_nyquist_limit(
         self, center_frequency, bandwidth_fraction, rolloff, zero_phase
     ):
         # Test that the filter doesn't raise an exception when
-        # center_freq + bandwidth / 2 is greater than the nyquist frequency
+        # center_freq + bandwidth / 2 is greater than the Nyquist frequency
 
         sample_rate = 16000
 
diff --git a/tests/test_room_simulator.py b/tests/test_room_simulator.py
index f2c1c125..f4a0d709 100644
--- a/tests/test_room_simulator.py
+++ b/tests/test_room_simulator.py
@@ -9,7 +9,7 @@
 
 
 def get_sinc_impulse(sample_rate, duration):
-    """Create a `duration` seconds chirp from 0Hz to `nyquist frequency`"""
+    """Create a `duration` seconds chirp from 0Hz to `Nyquist frequency`"""
     n = np.arange(-duration / 2, duration / 2, 1 / sample_rate)
 
     # Full band sinc impulse centered at half the duration