¿Cómo lidias con el recorte en situaciones de audio en vivo o en streaming?

While the article provides a comprehensive overview of clipping and its prevention, I'd like to emphasize the importance of understanding the equipment you're working with. Different gear has different thresholds, and while the general rule of keeping levels between -6dB and -3dB is sound, it's crucial to know the nuances of your specific setup. Moreover, in many recording situations, especially with voices that have a lot of dynamics, more than 6dB of headroom is advisable. Personally, I prefer to leave around 15dB of headroom. Additionally, in a live setting, it's not just about preventing clipping but also about ensuring the audio translates well in the venue's acoustics.

One of the most recommended approaches is always to prevent the problem from occurring as much as possible. In the podcast, for example, make sure you're recording the audio at a level ranging from -6db to -3db. The sound should not consistently exceed or fall below this range, and occasional variations might be acceptable in short parts. This will effectively prevent the problem.

Although a high gain setup is ideal to minimize a noise floor, the truth is that you cannot control a speaker's outbursts, laughter, mouth smacking, and other plosives that tend to clip. I usually use two light compressors in a series, but I first and foremost use very conservative gain to account for the unexpected. It's better to use a light gain and compensate with additional volume on a channel fader than to push the gain for the sake of noise floor and be constantly clipping.

The basics have already been discussed earlier, levels, gain stagging, compressors...As Far as I ve read, nobody mentionned a limiter (TG,Dynax,API...) there are tons), on the master bus or a good converter like the lavry gold will protect against any distorsion.

Adjusting gain is fundamental, but let's also discuss the quality of the preamps and the AD/DA converters. A poor-quality preamp can introduce noise even at optimal gain settings. How do you ensure that the entire signal path, from source to recording, maintains the highest fidelity?

Adjusting the gain level is not a fixed task. If you have specific settings, you should change them whenever necessary because, at times, especially in podcasts, you might need to alter the gain settings for the same person due to changes in their performance. Thus, there are no fixed gain settings.

It is important to have a comprehensive understanding of gain staging/structure. At each point in the signal chain you may typically have an input and an output attenuator, and the level passing through each of these points of control needs to be sent at a nominal level in order to avoid overloading the next stage of the signal flow. The exact level and reference point used to specify this amount of level will vary based on the specific piece of gear that you are utilizing. A few factors to consider based on your use case are: mic vs. line level, consumer vs. pro audio gear, speech or music, and the transparency or additive saturation (color) of the preamps in question.

While De-Clippers can assist in post-production, the advent of 32-bit float recording has been a game-changer in preventing clipping issues. With its vast dynamic range, it allows for post-recording gain adjustments without the risk of distortion. However, it's worth noting that while 32-bit offers more flexibility, it doesn't negate the importance of proper gain staging. How have your experiences been with 32-bit recording, and do you still find a need for De-Clippers in your workflow?

Effects and tools, in general, don't restore everything to its ideal state. Don't expect too much in this regard. Fortunately, some artificial intelligence tools can handle the problem better, such as Adobe Podcast. Therefore, I recommend using them in such cases.

Man sollte beachten, dass ein De-Clipper nur wirklich gut in der Post-Produktion funktioniert und nicht im Streaming-Prozess funktioniert. Es lohnt sich ihn zu kombinieren mit einem De-Crackle oder De-Click, wenn der De-Clip nicht das gewünschte Ergebnis erzielt hat.

While the article provides a comprehensive overview of clipping and its prevention, I'd like to emphasize the importance of understanding the equipment you're working with. Different gear has different thresholds, and while the general rule of keeping levels between -6dB and -3dB is sound, it's crucial to know the nuances of your specific setup. Moreover, in many recording situations, especially with voices that have a lot of dynamics, more than 6dB of headroom is advisable. Personally, I prefer to leave around 15dB of headroom. Additionally, in a live setting, it's not just about preventing clipping but also about ensuring the audio translates well in the venue's acoustics.

Oft entsteht Clipping nicht durch das eigentliche Signal, sondern Frequenz-Anteile, die im Audiosignal sind, aber gar nicht benötigt werden, z.B. Low-End Rumble oder auch hohes Rauschen über 10kHz. Es lohnt sich also immer einen Low-Cut und einen High-Cut auf jede Audiospur zu setzen, auch wenn man dieses kaum hört. Das Rauschen oder Rumpeln summiert sich gerne auf bei mehreren Spuren und ist dann ggf doch zu hören. In krassen Fällen, kann man sogar mit De-Noisern arbeiten, um die hohen Frequenzen nicht rauschen zu lassen.

Cutting low end out of a signal via high-pass filtering is an especially effective method of freeing up additional headroom, particularly in the context of live speech and achieving clarity. The human voice does not extend much farther down into that area of the spectrum than somewhere between 95-300Hz depending on the gender of the person. These low-mid and low frequency bands take up the majority of the headroom because their wavelength is so much longer and takes physical space in order to be accurately reproduced during the AD/DA conversion process. The same can be done on the high end, albeit less drastically because human hearing is much more sensitive to high frequency information. A healthy human hearing range roughly is 20Hz-20kHz.

While the text rightly emphasizes the importance of adjusting audio settings for different scenarios, it's crucial to remember that over-processing can degrade audio quality. It's always a balance between achieving the desired sound and maintaining the integrity of the original recording. I've often found that less is more when it comes to post-production. Also, while feedback from listeners is valuable, relying solely on it can be misleading. It's essential to trust your trained ears and use professional tools to make informed decisions.

In this specific issue, using tools might improve things to some extent, but they won't solve everything. Therefore, I still recommend using the newer AI tools as they provide better improvements compared to conventional tools used by sound engineers.

Familiarizing yourself with the equipment you are using is another important step in being able to optimize your system. As was mentioned, it is important to keep in mind the audience and environment that you will be presenting the audio in. Knowing how your audio should be delivered, whether for character or clarity, should be an important decision in how you process your signal. Not overdoing it is also important. As the processing capabilities of our computers and digital software increases, so does the temptation to overprocess our signals. Many times less can be more.

In general, this problem can be bothersome. So, make sure to avoid it because preventing it is much easier than trying to solve it after it occurs. Always ensure and monitor your recording settings. Recording the audio at levels between -6db and -3db will be significantly safe and will prevent the occurrence of that problem.