Single band compressor adobe audition как
Важной частью процесса обработки вокала является его компрессия — уменьшение динамического диапазона сигнала. Что это означает?
Во время записи вокала, одни его части звучат громче других. Всё это происходит по нескольким причинам. Первая – это отдаление и приближение к микрофону вокалиста в процессе пения. Вторая – разница в громкости при пении куплетов и припевов. Обычно припевы получаются более громкими из-за более эмоционального исполнения.
Обработка вокала с помощью инструмента Dynamics Processing
Для устранения разницы в громкости применяются специальные приборы динамической обработки звука, называемые компрессорами.
Рассмотрим процесс обработки вокала инструментом Dynamics Processing программы Adobe Audition 3.0
1. Запустим программу Adobe Audition 3.0 и загрузим наш вокал, выбрав в меню File / Open. На графике волновой формы вокала отчетливо видна разница в громкости на разных участках. Также мы можем увидеть превышение сигналом порога в 0 дБ, что чревато появлением клиппирования.
2. Используем для обработки вокала инструмент Dynamics Processing. Для этого выберем в меню Effects / Amplitude and Compression / Dynamics Processing.
3. Настраиваем плагин. Для этого давайте рассмотрим 4 его вкладки.
Вкладка Graphic используется для прорисовки графика компрессии.
По горизонтальной оси регулируется уровень входного сигнала, а по вертикальной – выходного. Инструмент может быть использован как компрессор, лимитер, экспандер или гейт.
На этой вкладке вы можете добавлять на график дополнительные точки и регулировать их положение по необходимости.
На вкладке Traditional настройки плагина представлены в более привычном виде. Здесь можно настроить порог срабатывания (Threshold), коэффициент компрессии (Ratio) и режим работы плагина для шести точек, а также уровень сигнала на выходе после компрессии.
Вкладка Attack / Release предназначена соответственно для настроек параметров атаки (скорости срабатывания) и восстановления (скорости выключения). Здесь также можно увеличить уровень выходного сигнала.
Вкладка Band Limiting предназначена для выбора диапазона компрессирования с помощью параметров Low Cutoff (отсечение сигнала, который находится ниже заданного порога) и High Cutoff (отсечение сигнала, который находится выше заданного порога).
Для правильной настройки компрессора установите следующие значения: Threshold – 0 дБ, Ratio – 2, Attack – 2-10 мс, Release – 150 мс. Начинайте опускать порог срабатывания (Threshold) пока не почувствуете, что сигнал подвергается обработке. После этого, начинайте подбирать соответствующую степень сжатия с помощью параметра Ratio. Это значение колеблется в пределах 2:1 – 8:1. Большие значения могут сделать вокал сухим и безжизненным. Будьте аккуратны, постоянно сравнивайте обработанный сигнал с необработанным.
Если вы услышите так называемую «накачку» (один из нежелательных артефактов) сигнала в промежутках между фразами, попробуйте увеличить время восстановления или поднять порог срабатывания. Если теряется артикуляция отдельных слов, отрегулируйте атаку.
Я выставил следующие параметры:
Threshold — -17, 5
Ratio – 3, 98: 1
Attack – 8 мс
Release – 270 мс
Output Gain – 5 дБ
Вы можете увидеть эти параметры на рисунках выше.
4. После выполнения необходимых настроек нажимаем Ок.
Давайте сравним вокал до обработки и после.
Послушаем голос до обработки.
Послушаем голос после обработки
Как видим динамический диапазон стал намного уже. Теперь весь вокал имеет приблизительно одинаковый уровень громкости.
Хочется отметить, что для компрессии вы можете использовать более удобные и привычные для вас компрессоры, например Waves C1, Tube-modeled Compressor программы Adobe Audition и многие другие.
На этом этап обработки вокала с помощью компрессии можно считать завершённым.
В следующей статье поговорим о коррекции вокала с помощью инструмента Melodyne.
Наверняка вы видели видеоролики, или слушали аудиозаписи, в которых звук распределялся неравномерно. Например, когда какой-нибудь стример говорит тихо, да так, что приходится делать звук, на вашем устройстве, громче, но в какой-то момент – что-то происходит, и он начинает орать как сумасшедший. Вроде бы ничего такого, но если ты живешь не один, и если смотришь это видео ночью – то совсем неприятно будить своих домашних такой ерундой.
К сожалению, не многие стримеры, ютуберы и т.д. обращают внимание на потребности своей аудитории. Им все равно на удобства пользователей и из-за этого – они теряют часть аудитории. Для этого мы и делаем статью, в которой расскажем, как выровнять громкость в Adobe Audition.
На самом деле – все очень просто. Для примера – возьмем аудиозапись с криками школьника из майнкрафта.
К сожалению, передать громкость этих воплей на ультра высоких дБ – мы вам не сможем, но по ползунку на аудиодорожке – вы должны понять, что там происходит. Пусть остальное доработает фантазия. Итак, нам нужно выровнять эти вопли.
Для этого выделяем нашу запись. После этого – заходим в пункт «эффекты», расположенную вверху. Там мы находим пункт Amplitude and compression, и в новой табличке – выбираем Single-band Cpmpression.
Далее – у вас появится маленькая табличка, с 5 ползунками. Чтобы это все не размусоливать – просто сделайте все так, как показано на скрине ниже.
Нажимаете Apply. Далее – вы видите, что наша аудиозапись – осталась прежней, но высокие горки воплей – заметно поутихли, застыв на определенном уровне громкости.
Вот, собственно, и все. Таким образом вы сможете редактировать звук любой вашей аудиозаписи и любого видео, а главное – вы будете знать, что ваши подписчики не будут вас ругать, за очередные вопли, во время убийства или смерти, на вашем видео. Заметьте, что все звуки остаются на своих местах, просто их громкость останавливается на определенном уровне.
The Amplitude And Compression > Amplify effect boosts or attenuates an audio signal. Because the effect operates in real time, you can combine it with other effects in the Effects Rack.
Boost or attenuate individual audio channels.
Moves the channel sliders together.
The Amplitude and Compression > Channel Mixer effect alters the balance of stereo or surround channels. You change the apparent position of sounds, correct mismatched levels, or address phasing issues.
Select the output channel.
Input channel sliders
To mix into the output channel, determine the percentage of the current channels. For a stereo file, for example, an L value of 50 and an R value of 50 results in an output channel that contains equal audio from the current left and right channels.
Inverts a channel’s phase. (To understand this key audio concept, see How sound waves interact.) Inverting all channels causes no perceived difference in sound. Inverting only one channel, however, can greatly change the sound.
The Amplitude and Compression > DeEsser effect removes sibilance, “ess” sounds heard in speech and singing that can distort high frequencies.
The graph reveals the processed frequencies. To see how much audio content exists in the processed range, click the Preview button .
Choose Broadband to uniformly compress all frequencies or Multiband to only compress the sibilance range. Multiband is best for most audio content but slightly increases processing time.
Sets the amplitude above which compression occurs.
Specifies the frequency at which sibilance is most intense. To verify, adjust this setting while playing audio.
Determines the frequency range that triggers the compressor.
To visually adjust Center Frequency and Bandwidth, drag the edges of the selection in the graph.
Output Sibilance Only
Lets you hear detected sibilance. Start playback, and fine-tune settings above.
Shows the compression level of the processed frequencies.
The Amplitude And Compression > Dynamics Processing effect can be used as a compressor, limiter, or expander. As a compressor and limiter, this effect reduces dynamic range, producing consistent volume levels. As an expander, it increases dynamic range by reducing the level of low‑level signals. (With extreme expander settings, you can create a noise gate that totally eliminates noise below a specific amplitude threshold.)
The Dynamics Processing effect can produce subtle changes that you notice only after repeated listening. When applying this effect in the Waveform Editor, use a copy of the original file so you can return to the original audio if necessary.
In the Dynamic Processing Effect , you can view the Level Meter and the Gain Reduction Meter. Level Mete r shows the input level of the audio and Gain Reduction Meter shows how audio signals are compressed or expanded. These meters are visible on the right side of the graph as shown below.
Use the Broadcast Limiter preset to simulate the processed sound of a contemporary radio station.
Dynamics tab
Depicts input level along the horizontal ruler (x‑axis) and the new output level along the vertical ruler (y‑axis). The default graph, with a straight line from the lower left to the upper right, depicts a signal that has been left untouched; every input level has the same output level. Adjusting the graph changes the relationship between input and output levels, altering dynamic range.
For example, if a desirable sonic element occurs around ‑20 dB, you can boost the input signal at that level, but leave everything else unchanged. You can also draw an inverse line (from the upper left to the lower right) that boosts quiet sounds and suppress loud ones.
Adds control point in graph using numerical input and output levels you specify. This method is more precise than clicking the graph to add points.
To numerically adjust an existing control point, right-click it, and choose Edit Point.
Removes selected point from the graph.
Flips the graph, converting compression into expansion, or the other way around.
Note: You can invert a graph only if it has points in the two default corners (‑100, ‑100 and 0, 0) if its output level increases from left to right (that is, each control point must be higher than the one to its left).
Resets the graph to its default state.
Creates smoother, curved transitions between control points, rather than more abrupt, linear transitions. (See About spline curves for graphs.)
Boosts the processed signal.
Settings tab
Provides overall settings.
Addresses transient spikes that can occur at the onset of loud signals that extend beyond the compressor’s Attack Time settings. Extending Look-Ahead Time causes compression to attack before the audio gets loud, ensuring that amplitude never exceeds a certain level. Conversely, reducing Look-Ahead Time is desirable to enhance the impact of percussive music like drum hits.
Completely silences signals that are expanded below a 50-to-1 ratio.
Determines the original input amplitude.
Applies gain to the signal before it enters the Level Detector.
Determines how many milliseconds it takes for the input signal to register a changed amplitude level. For example, if audio suddenly drops 30 dB, the specified attack time passes before the input registers an amplitude change. This avoids erroneous amplitude readings due to temporary changes.
Determines how many milliseconds the current amplitude level is maintained before another amplitude change can register.
Use fast attack and release settings for audio with fast transients, and slower settings for less percussive audio.
Determines levels based on amplitude peaks. This mode is a bit more difficult to use than RMS, because peaks aren’t precisely reflected in the Dynamics graph. However, it can be helpful when audio has loud transient peaks you want to subdue.
Determines levels based on the root-mean-square formula, an averaging method that more closely matches the way people perceive volume. This mode precisely reflects amplitudes in the Dynamics graph. For example, a limiter (flat horizontal line) at ‑10 dB reflects an average RMS amplitude of ‑10 dB.
Amplifies or attenuates the signal depending on the amplitude detected.
Applies gain to the output signal after all dynamics processing.
Determines how many milliseconds it takes for the output signal to reach the specified level. For example, if audio suddenly drops 30 dB, the specified attack time passes before the output level changes.
Determines how many milliseconds the current output level is maintained.
Note: If the sum of Attack and Release times is too short (less than about 30 milliseconds), audible artifacts can be heard. To see good attack and release times for different types of audio content, choose various options from the Presets menu.
Processes all channels equally, preserving the stereo or surround balance. For example, a compressed drum beat on the left channel reduces the right channel level by an equal amount.
Restricts dynamics processing to a specific frequency range.
Is the lowest frequency that dynamics processing affects.
Is the highest frequency that dynamics processing affects.
The Amplitude and Compression > Dynamics Effects consist of four sections. They are Auto Gate, Compressor, Expander, and Limiter. You can individually control each one of the section. The LED and gain reduction meters helps you get the overview about how the audio signal is processed.
The different parameters under Dynamic Effects are as follows:
• Auto Gate: Removes noise below a certain amplitude threshold. The LED meter is green when audio passes through the gate. The meter turns red when there is no audio passing, and yellow during the attack, release, and hold times.
• Compressor: Reduces the dynamic range of the audio signal by attenuating audio that exceeds a specific threshold. The Ratio parameter can be used control the change in dynamic range while Attack and Release parameter changes the temporal behavior. Use the Gain parameter to increase the audio level after compressing the signal. The Gain Reduction meter shows how much the audio level is reduced.
• Expander: Increases the dynamic range of the audio signal by attenuating audio below the specified threshold. The ratio parameter can be used to control the change in dynamic range. The gain reduction meter shows the level of reduction in audio level.
• Limiter: Attenuate audio that exceeds a specified threshold. The meter LED turns on when the signal is limited.
To reduce amplitude by varying amounts over time, choose Fade Envelope ( Effects > Amplitude and Compression ).
In the Waveform Editor panel, click the yellow envelope line to add keyframes, and drag them up or down to change amplitude. To quickly select, reposition, or delete multiple keyframes, see Adjust automation with keyframes.
To create smoother, curved transitions between keyframes, rather than linear transition, select the Spline Curves option. See About spline curves for graphs.
To boost or reduce amplitude over time, choose Gain Envelope ( Effects > Amplitude and Compression ).
In the Waveform Editor panel, click the yellow envelope line to add keyframes, and drag them up or down to change amplitude. To quickly select, reposition, or delete multiple keyframes, see Adjust automation with keyframes.
To create smoother, curved transitions between keyframes, rather than linear transitions, select the Spline Curves option t. See About spline curves for graphs.
The Amplitude And Compression > Hard Limiter effect greatly attenuates audio that rises above a specified threshold. Typically, limiting is applied with an input boost, a technique that increases overall volume while avoiding distortion.
Sets the maximum sample amplitude allowed.
Tip: To avoid clipping when working with 16‑bit audio, set this value to no more than ‑0.3 dB. If you set it even lower, to ‑3 dB, you’ll have a little more clearance for any future edits.
Preamplifies audio before you limit it, making a selection louder without clipping it. As you increase this level, compression increases. Try extreme settings to achieve the loud, high‑impact audio heard in contemporary pop music.
Sets the amount of time (in milliseconds) to attenuate the audio before the loudest peak is hit.
Note: Make sure that the value is at least 5 milliseconds. If this value is too small, audible distortion effects occur.
Sets the time (in milliseconds) for the attenuation to rebound back 12 dB (or roughly the time needed for audio to resume normal volume if a loud peak is encountered). In general, a setting of around 100 (the default) works well and preserves low bass frequencies.
Note: If this value is too large, audio can remain quiet and not resume normal levels for a while.
Links the loudness of all channels together, preserving the stereo or surround balance.
The Amplitude And Compression > Multiband Compressor effect lets you independently compress four different frequency bands. Because each band typically contains unique dynamic content, multiband compression is a powerful tool for audio mastering.
Controls in the Multiband Compressor let you precisely define crossover frequencies and apply band‑specific compression settings. To preview bands in isolation, or Bypass buttons to pass bands through without processing, click Solo buttons. After you fine‑tune individual bands, select Link Band Controls to adjust them globally, and then optimize overall volume with the Output Gain slider and Limiter settings.
To change compression settings over time, use automation lanes in the Multitrack Editor. (See Automating track settings.)
A. Frequency bands B. Crossover markers C. Bypassed band (no processing applied) D. Amplitude scale E. Frequency scale
Sets the crossover frequencies, which determine the width of each band. Either enter specific Low, Midrange, and High frequencies, or drag the crossover markers above the graph.
A. Solo B. Bypass C. Threshold slider D. Input Level meters E. Gain Reduction meters
Let you hear specific frequency bands. Enable one Solo button at a time to hear bands in isolation, or enable multiple buttons to hear two or more bands together.
Bypass individual bands so they pass through without processing.
Alt‑click (Windows) or Option-click (Mac OS) Solo or Bypass buttons to quickly apply a unique setting to one band.
Set the input level at which compression begins. Possible values range from ‑60 to 0 dB. The best setting depends on audio content and musical style. To compress only extreme peaks and retain more dynamic range, try thresholds around 5 dB below the peak input level. To highly compress audio and greatly reduce dynamic range, try settings around 15 dB below the peak input level.
Input Level meters
Measure input amplitude. To reset peak and clip indicators, double‑click the meters.
Gain Reduction meters
Measure amplitude reductions with red meters that extend from top (minimal reduction) to bottom (maximum reduction).
Boosts or cuts amplitude after compression. Possible values range from ‑18 to +18 dB, where 0 is unity gain.
Sets a compression ratio between 1‑to‑1 and 30‑to‑1. For example, a setting of 3.0 outputs 1 dB for every 3-dB increase above the compression threshold. Typical settings range from 2.0 to 5.0; higher settings produce the compressed sound often heard in pop music.
Determines how quickly compression is applied when audio exceeds the threshold. Possible values range from 0 milliseconds to 500 milliseconds. The default, 10 milliseconds, works well for a wide range of audio. Faster settings work better for audio with fast transients, but such settings sound unnatural for less percussive audio.
Determines how quickly compression stops after audio drops below the threshold. Possible values range from 0 milliseconds to 5000 milliseconds. The default, 100 milliseconds, works well for a wide range of audio. Try faster settings for audio with fast transients, and slower settings for less percussive audio.
Boosts or cuts overall output level after compression. Possible values range from ‑18 to +18 dB, where 0 is unity gain. To reset peak and clip indicators, double‑click the meters.
Applies limiting after Output Gain, at the end of the signal path, optimizing overall levels. Specify Threshold, Attack, and Release settings that are less aggressive than similar band‑specific settings. Then specify a Margin setting to determine the absolute ceiling relative to 0 dBFS.
Tip: To create compressed audio, enable the Limiter, and then experiment with high Output Gain settings.
Spectrum On Input
Displays the frequency spectrum of the input signal, rather than the output signal, in the multiband graph. To quickly see the amount of compression applied to each band, toggle this option on and off.
Applies immediate, hard limiting at the current Margin setting. (Deselect this option to apply slower soft limiting, which sounds less compressed but can exceed the Margin setting.)
Note: The maximum Attack time for brickwall limiting is 5 ms.
Link Band Controls
Lets you globally adjust the compression settings for all bands, while retaining relative differences between bands.
To temporarily link band controls, hold down Alt+Shift (Windows) or Option+Shift (Mac OS). To reset a control in all bands, hold down Ctrl+Alt+Shift (Windows) or Command+Option+Shift (Mac OS), and click the control.
This effect requires offline processing. While it is open, you cannot edit the waveform, adjust selections, or move the current-time indicator.
The Amplitude And Compression > Normalize effect lets you set a peak level for a file or selection. When you normalize audio to 100%, you achieve the maximum amplitude that digital audio allows—0 dBFS. If you’re sending audio to a mastering engineer, however, normalize audio to between –3 and –6 dBFS, providing a cushion for further processing.
The Normalize effect amplifies the entire file or selection equally. For example, if the original audio reaches a loud peak of 80% and a quiet low of 20%, normalizing to 100% amplifies the loud peak to 100% and the quiet low to 40%.
To apply RMS normalization, choose Effects > Match Volume. If desired, you can apply that command to only one file. (See Match volume across multiple files.)
The Amplitude And Compression > Amplify effect boosts or attenuates an audio signal. Because the effect operates in real time, you can combine it with other effects in the Effects Rack.
Boost or attenuate individual audio channels.
Moves the channel sliders together.
The Amplitude and Compression > Channel Mixer effect alters the balance of stereo or surround channels. You change the apparent position of sounds, correct mismatched levels, or address phasing issues.
Select the output channel.
Input channel sliders
To mix into the output channel, determine the percentage of the current channels. For a stereo file, for example, an L value of 50 and an R value of 50 results in an output channel that contains equal audio from the current left and right channels.
Inverts a channel’s phase. (To understand this key audio concept, see How sound waves interact.) Inverting all channels causes no perceived difference in sound. Inverting only one channel, however, can greatly change the sound.
The Amplitude and Compression > DeEsser effect removes sibilance, “ess” sounds heard in speech and singing that can distort high frequencies.
The graph reveals the processed frequencies. To see how much audio content exists in the processed range, click the Preview button .
Choose Broadband to uniformly compress all frequencies or Multiband to only compress the sibilance range. Multiband is best for most audio content but slightly increases processing time.
Sets the amplitude above which compression occurs.
Specifies the frequency at which sibilance is most intense. To verify, adjust this setting while playing audio.
Determines the frequency range that triggers the compressor.
To visually adjust Center Frequency and Bandwidth, drag the edges of the selection in the graph.
Output Sibilance Only
Lets you hear detected sibilance. Start playback, and fine-tune settings above.
Shows the compression level of the processed frequencies.
The Amplitude And Compression > Dynamics Processing effect can be used as a compressor, limiter, or expander. As a compressor and limiter, this effect reduces dynamic range, producing consistent volume levels. As an expander, it increases dynamic range by reducing the level of low‑level signals. (With extreme expander settings, you can create a noise gate that totally eliminates noise below a specific amplitude threshold.)
The Dynamics Processing effect can produce subtle changes that you notice only after repeated listening. When applying this effect in the Waveform Editor, use a copy of the original file so you can return to the original audio if necessary.
In the Dynamic Processing Effect , you can view the Level Meter and the Gain Reduction Meter. Level Mete r shows the input level of the audio and Gain Reduction Meter shows how audio signals are compressed or expanded. These meters are visible on the right side of the graph as shown below.
Use the Broadcast Limiter preset to simulate the processed sound of a contemporary radio station.
Dynamics tab
Depicts input level along the horizontal ruler (x‑axis) and the new output level along the vertical ruler (y‑axis). The default graph, with a straight line from the lower left to the upper right, depicts a signal that has been left untouched; every input level has the same output level. Adjusting the graph changes the relationship between input and output levels, altering dynamic range.
For example, if a desirable sonic element occurs around ‑20 dB, you can boost the input signal at that level, but leave everything else unchanged. You can also draw an inverse line (from the upper left to the lower right) that boosts quiet sounds and suppress loud ones.
Adds control point in graph using numerical input and output levels you specify. This method is more precise than clicking the graph to add points.
To numerically adjust an existing control point, right-click it, and choose Edit Point.
Removes selected point from the graph.
Flips the graph, converting compression into expansion, or the other way around.
Note: You can invert a graph only if it has points in the two default corners (‑100, ‑100 and 0, 0) if its output level increases from left to right (that is, each control point must be higher than the one to its left).
Resets the graph to its default state.
Creates smoother, curved transitions between control points, rather than more abrupt, linear transitions. (See About spline curves for graphs.)
Boosts the processed signal.
Settings tab
Provides overall settings.
Addresses transient spikes that can occur at the onset of loud signals that extend beyond the compressor’s Attack Time settings. Extending Look-Ahead Time causes compression to attack before the audio gets loud, ensuring that amplitude never exceeds a certain level. Conversely, reducing Look-Ahead Time is desirable to enhance the impact of percussive music like drum hits.
Completely silences signals that are expanded below a 50-to-1 ratio.
Determines the original input amplitude.
Applies gain to the signal before it enters the Level Detector.
Determines how many milliseconds it takes for the input signal to register a changed amplitude level. For example, if audio suddenly drops 30 dB, the specified attack time passes before the input registers an amplitude change. This avoids erroneous amplitude readings due to temporary changes.
Determines how many milliseconds the current amplitude level is maintained before another amplitude change can register.
Use fast attack and release settings for audio with fast transients, and slower settings for less percussive audio.
Determines levels based on amplitude peaks. This mode is a bit more difficult to use than RMS, because peaks aren’t precisely reflected in the Dynamics graph. However, it can be helpful when audio has loud transient peaks you want to subdue.
Determines levels based on the root-mean-square formula, an averaging method that more closely matches the way people perceive volume. This mode precisely reflects amplitudes in the Dynamics graph. For example, a limiter (flat horizontal line) at ‑10 dB reflects an average RMS amplitude of ‑10 dB.
Amplifies or attenuates the signal depending on the amplitude detected.
Applies gain to the output signal after all dynamics processing.
Determines how many milliseconds it takes for the output signal to reach the specified level. For example, if audio suddenly drops 30 dB, the specified attack time passes before the output level changes.
Determines how many milliseconds the current output level is maintained.
Note: If the sum of Attack and Release times is too short (less than about 30 milliseconds), audible artifacts can be heard. To see good attack and release times for different types of audio content, choose various options from the Presets menu.
Processes all channels equally, preserving the stereo or surround balance. For example, a compressed drum beat on the left channel reduces the right channel level by an equal amount.
Restricts dynamics processing to a specific frequency range.
Is the lowest frequency that dynamics processing affects.
Is the highest frequency that dynamics processing affects.
The Amplitude and Compression > Dynamics Effects consist of four sections. They are Auto Gate, Compressor, Expander, and Limiter. You can individually control each one of the section. The LED and gain reduction meters helps you get the overview about how the audio signal is processed.
The different parameters under Dynamic Effects are as follows:
• Auto Gate: Removes noise below a certain amplitude threshold. The LED meter is green when audio passes through the gate. The meter turns red when there is no audio passing, and yellow during the attack, release, and hold times.
• Compressor: Reduces the dynamic range of the audio signal by attenuating audio that exceeds a specific threshold. The Ratio parameter can be used control the change in dynamic range while Attack and Release parameter changes the temporal behavior. Use the Gain parameter to increase the audio level after compressing the signal. The Gain Reduction meter shows how much the audio level is reduced.
• Expander: Increases the dynamic range of the audio signal by attenuating audio below the specified threshold. The ratio parameter can be used to control the change in dynamic range. The gain reduction meter shows the level of reduction in audio level.
• Limiter: Attenuate audio that exceeds a specified threshold. The meter LED turns on when the signal is limited.
To reduce amplitude by varying amounts over time, choose Fade Envelope ( Effects > Amplitude and Compression ).
In the Waveform Editor panel, click the yellow envelope line to add keyframes, and drag them up or down to change amplitude. To quickly select, reposition, or delete multiple keyframes, see Adjust automation with keyframes.
To create smoother, curved transitions between keyframes, rather than linear transition, select the Spline Curves option. See About spline curves for graphs.
To boost or reduce amplitude over time, choose Gain Envelope ( Effects > Amplitude and Compression ).
In the Waveform Editor panel, click the yellow envelope line to add keyframes, and drag them up or down to change amplitude. To quickly select, reposition, or delete multiple keyframes, see Adjust automation with keyframes.
To create smoother, curved transitions between keyframes, rather than linear transitions, select the Spline Curves option t. See About spline curves for graphs.
The Amplitude And Compression > Hard Limiter effect greatly attenuates audio that rises above a specified threshold. Typically, limiting is applied with an input boost, a technique that increases overall volume while avoiding distortion.
Sets the maximum sample amplitude allowed.
Tip: To avoid clipping when working with 16‑bit audio, set this value to no more than ‑0.3 dB. If you set it even lower, to ‑3 dB, you’ll have a little more clearance for any future edits.
Preamplifies audio before you limit it, making a selection louder without clipping it. As you increase this level, compression increases. Try extreme settings to achieve the loud, high‑impact audio heard in contemporary pop music.
Sets the amount of time (in milliseconds) to attenuate the audio before the loudest peak is hit.
Note: Make sure that the value is at least 5 milliseconds. If this value is too small, audible distortion effects occur.
Sets the time (in milliseconds) for the attenuation to rebound back 12 dB (or roughly the time needed for audio to resume normal volume if a loud peak is encountered). In general, a setting of around 100 (the default) works well and preserves low bass frequencies.
Note: If this value is too large, audio can remain quiet and not resume normal levels for a while.
Links the loudness of all channels together, preserving the stereo or surround balance.
The Amplitude And Compression > Multiband Compressor effect lets you independently compress four different frequency bands. Because each band typically contains unique dynamic content, multiband compression is a powerful tool for audio mastering.
Controls in the Multiband Compressor let you precisely define crossover frequencies and apply band‑specific compression settings. To preview bands in isolation, or Bypass buttons to pass bands through without processing, click Solo buttons. After you fine‑tune individual bands, select Link Band Controls to adjust them globally, and then optimize overall volume with the Output Gain slider and Limiter settings.
To change compression settings over time, use automation lanes in the Multitrack Editor. (See Automating track settings.)
A. Frequency bands B. Crossover markers C. Bypassed band (no processing applied) D. Amplitude scale E. Frequency scale
Sets the crossover frequencies, which determine the width of each band. Either enter specific Low, Midrange, and High frequencies, or drag the crossover markers above the graph.
A. Solo B. Bypass C. Threshold slider D. Input Level meters E. Gain Reduction meters
Let you hear specific frequency bands. Enable one Solo button at a time to hear bands in isolation, or enable multiple buttons to hear two or more bands together.
Bypass individual bands so they pass through without processing.
Alt‑click (Windows) or Option-click (Mac OS) Solo or Bypass buttons to quickly apply a unique setting to one band.
Set the input level at which compression begins. Possible values range from ‑60 to 0 dB. The best setting depends on audio content and musical style. To compress only extreme peaks and retain more dynamic range, try thresholds around 5 dB below the peak input level. To highly compress audio and greatly reduce dynamic range, try settings around 15 dB below the peak input level.
Input Level meters
Measure input amplitude. To reset peak and clip indicators, double‑click the meters.
Gain Reduction meters
Measure amplitude reductions with red meters that extend from top (minimal reduction) to bottom (maximum reduction).
Boosts or cuts amplitude after compression. Possible values range from ‑18 to +18 dB, where 0 is unity gain.
Sets a compression ratio between 1‑to‑1 and 30‑to‑1. For example, a setting of 3.0 outputs 1 dB for every 3-dB increase above the compression threshold. Typical settings range from 2.0 to 5.0; higher settings produce the compressed sound often heard in pop music.
Determines how quickly compression is applied when audio exceeds the threshold. Possible values range from 0 milliseconds to 500 milliseconds. The default, 10 milliseconds, works well for a wide range of audio. Faster settings work better for audio with fast transients, but such settings sound unnatural for less percussive audio.
Determines how quickly compression stops after audio drops below the threshold. Possible values range from 0 milliseconds to 5000 milliseconds. The default, 100 milliseconds, works well for a wide range of audio. Try faster settings for audio with fast transients, and slower settings for less percussive audio.
Boosts or cuts overall output level after compression. Possible values range from ‑18 to +18 dB, where 0 is unity gain. To reset peak and clip indicators, double‑click the meters.
Applies limiting after Output Gain, at the end of the signal path, optimizing overall levels. Specify Threshold, Attack, and Release settings that are less aggressive than similar band‑specific settings. Then specify a Margin setting to determine the absolute ceiling relative to 0 dBFS.
Tip: To create compressed audio, enable the Limiter, and then experiment with high Output Gain settings.
Spectrum On Input
Displays the frequency spectrum of the input signal, rather than the output signal, in the multiband graph. To quickly see the amount of compression applied to each band, toggle this option on and off.
Applies immediate, hard limiting at the current Margin setting. (Deselect this option to apply slower soft limiting, which sounds less compressed but can exceed the Margin setting.)
Note: The maximum Attack time for brickwall limiting is 5 ms.
Link Band Controls
Lets you globally adjust the compression settings for all bands, while retaining relative differences between bands.
To temporarily link band controls, hold down Alt+Shift (Windows) or Option+Shift (Mac OS). To reset a control in all bands, hold down Ctrl+Alt+Shift (Windows) or Command+Option+Shift (Mac OS), and click the control.
This effect requires offline processing. While it is open, you cannot edit the waveform, adjust selections, or move the current-time indicator.
The Amplitude And Compression > Normalize effect lets you set a peak level for a file or selection. When you normalize audio to 100%, you achieve the maximum amplitude that digital audio allows—0 dBFS. If you’re sending audio to a mastering engineer, however, normalize audio to between –3 and –6 dBFS, providing a cushion for further processing.
The Normalize effect amplifies the entire file or selection equally. For example, if the original audio reaches a loud peak of 80% and a quiet low of 20%, normalizing to 100% amplifies the loud peak to 100% and the quiet low to 40%.
To apply RMS normalization, choose Effects > Match Volume. If desired, you can apply that command to only one file. (See Match volume across multiple files.)
Освежив свою память, и немного покапавшись в теории, я почти решил свой вопрос, ответ на который вкратце опишу в начале данного топика.
Итак, задача состоит в нормализации записанного голоса с помощью программы Adobe Auditon.
Зачастую, записанный на камеру либо микрофон звук, имеет пиковые всплески и заглушённые места, которые для лучшего восприятия следует уровнять.
Для лучшего понимания о чём я говорю, можно вспомнить разрывающую колонки рекламу по телевизору, которая очень часто воспроизводится намного громче основного видеоконтента.
На данном графике можно наблюдать такие пиковые всплески а также приглушённые места:
Открываем Single-band Compressor. Я буду показывать на его примере. Остальные компрессоры выполняют почти такую же функцию, только различны в оформлении и могут иметь дополнительные настройки.
И теперь, для понимания настроек, дам их краткое описание своими словами:
Threshold (порог) - это порог срабатывания компрессора. т.е., устанавливая его значение на определенный уровень (например -21 dB), мы ему показываем примерный уровень, где наш голос/звук ровный/стабильный. Иными словами, показываем компрессору границу, выше которой звук будет обрезаться.
Ratio (соотношение) - это значение, показывающее во сколько раз подрезать звук, указанный в Threshold. Если поставить значение 30 (максимальное), мы покажем компрессору, что звук, который выше -21 dB, следует понизить в 30 раз. Это жёсткое обрезание, которое я ставлю почти всегда по-умолчанию.
Attack (атака) - скорость срабатывания компрессора, т.е. время через которое идёт данная подрезка/усиление звука. Я ставлю всегда 0 (ноль), дабы небыло что-то типа постепенного затухания/плавного увеличения звука.
Release (восстановление) - время через которое компрессор прекратит обрабатывать данный пик звука. Я устанавливаю максимум 5000 ms, что даёт очень жёсткую обрезку.
Gain/Output Gain (усиление выхода) - повышает или понижает общий уровень звука после сжатия. В нашем случае можно сказать, что это значение, определяющее до скольки нужно поднять тихие звуки. Я ставлю всегда прим. на 3 dB ниже значения Threshold только с плюсовым значением.
Обработав данный звуковой файл с такими настройками:
В итоге мы получаем довольно ровный звук на общем уровне -3 dB
Теперь его можно и оставить на этом значении, либо подтянуть до нужного обычным регулятором уровня звука
Как-то так. Конечно это очень грубая обработка, однако тоже довольно неплохо в итоге получается.
З.Ы. Надеюсь кому-то это будет полезно)))
З.Ы.З.Ы. Писал быстро, на голодный желудок, поэтому могут быть ошибки и неточности. Звыняйтэ если шо)))
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