Re: Convert Video Youtube Jadi Audio Mp3

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Kian Trip

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Jul 16, 2024, 6:16:14 AM7/16/24
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Instavideosave.net offers a free instagram to mp3 converter tool which allows you to extract and downloadMp3 Audio from Instagram Reels, Igtv, videos. it's a quick and easy way to convert and downloadinstagram videos to Mp3 Audio online.

In order to download instagram reels, igtv, video to audio mp3, first copy the link to the video and thenpaste it into the input form on instavideosave.net/audio/. Then after, click download button to convert anddownload instagram videos, igtv, reels to mp3 audio only.

convert video youtube jadi audio mp3


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instagram reels is one of the most popular shorts video creating plateform, thousand of users post reels oninstagram from which most of users want to use a particular audio in their videos. Using instagram video toaudio downloader, you can convert reels to mp3 and download audio from a reel video by link.

I know cubase 13 has the facility to easily convert mono to stereo and this is great, but the GUI is too painful for me old eyes. so, in cubase 12, it seems you have 2 options. 1 - create a stereo track and copy and paste all the audio from the mono track. 2 - route the mono track to a stereo group channel.

channel A - mono audio file playing back as a mono channel in cubase mixer
channel B - mono audio file playing back as a stereo channel in cubase mixer
all audio parts were copied and pasted across. project pan law is 4.5.

if you pan both hard left or right, channel B is 4.5db lower in volume, despite sounding the same volume when panned centrally. put boz pan knob 2 on channel B set to 4.5db pan law, panning from within pan knob 2 now results in same apparent volume.

a further problem appears to be the panning issue. if you want a mono track to be a stereo channel, the panning on the stereo channel, whether its a stereo group with the original mono channel routed to it, or a new stereo channel with all the data copied and pasted across, will not behave the same. the panner on the stereo channel will not have the same pan law as the original mono channel as set in project settings, resulting in a lower level when panned hard LR on the stereo channel. I use 4.5db pan law generally, and when routing mono audio to a stereo channel in these ways, the pan law on that stereo channel is not 4.5db. the only way for it to be 4.5 again on that channel is to use, for example, boz pan knob set at 4.5db panlaw inbstead of the cubase panner.

frankly this is all pretty weird, and potentially affects no one at all in real world use if you use your ears and start the mix with the routing already set up and the mono tracks already converted to stereo in whichever way you choose. good to know tho. if you convert a mono track to stereo mid mix and its panned at all, it could catch you out.

Panorama for mono, on the other hand, regulates how the power of the signal is spread on the stereo spectrum, ie. the sum will always be 100%. If panorama is in the center the left channel gets 50% (that is -6dB) and the right channel likewise. The sum of both channels is 100%. If you set it to full left the left channel has 100% and the right 0% (sum is 100%).
Here is where the Pan Law chirps in. You can use Pan Law settings to NOT have the sum of a mono signal = 100%. It works only for mono channels.

frankly this is all pretty weird, and potentially affects no one at all in real world use if you use your ears and start the mix with the routing already set up and the mono tracks already converted to stereo in whichever way you choose.

However the Pan Law is more important for a signal moving around in the Panorama. Try this- take a mono signal routed to a Stereo Group so you can Pan the signal. Create Automation so the signal Pans back & forth from left to right. Now listen to it with different settings for the Pan Law.

The answers on this question have since become a bit of a mess, with many containing redundant information and others complete inaccuracies. This answer is an attempt to streamline the information in these answers while doing away with the problems in them.

The -ac 2 switch works by mixing proportions of the first 5 channels from the source's 6-channel stream - Back Left, Back Right, Front Left, Front Right and Front Center - into the Front Left and Front Right channels of the output stereo stream:

Unfortunately, in my tests -ac 2 resulted in overall levels of both music and dialogue that were the most different to the source, making it the downmix formula that gives the worst output out of all the formulae I tested, although you may test it and find that it gives you a perfectly adequate downmix for your needs, in which case using any other formula would be overkill for you.

As pointed out by Mephisto in his answer, if the dialogue and the music sound well-balanced among each other to you but simply lack volume, you can downmix the stream while also increasing its volume:

For the -vol switch, 100% volume in the source is equivalent to the integer value 256, and using a larger value than this will increase the overall volume of the audio stream. However, note that doing so too much may result in distortion or artifacts, especially during its louder sections.

If you want a more high quality downmix, or you absolutely must include the LFE stream into your output, you can use FFmpeg's audio filter switch (-af) to downmix the audio using a custom mix formula.

As of the time of posting this answer, the top-voted answer to this question was Gregory's, which puts the formula from the ATSC specification (see section 7.8.2, Downmixing into Two Channels) into an FFmpeg audio filter. This specification is itself directly linked to by the FFmpeg documentation on the topic, indicating it's highly likely to be the same formula that FFmpeg already implements for its -ac 2 switch. If this is true, then typing out the entire formula in Gregory's answer would be no different than using the -ac 2 switch, and therefore a waste of time.

It therefore seems pretty conclusive that the ATSC formula detailed in Gregory's answer is the same one already implemented by FFmpeg, and that using it is entirely redundant when it does nothing that -ac 2 doesn't, and is a much more cumbersome command.

Of the several included in the answers, this is the only one of the downmix formulae that appears to mix the LFE channel into the output stereo instead of discarding it entirely, and as a result, the one that ensures the least sound from the source is lost.

The overall volume level is higher and fuller than doing -ac 2, but also still lower than the below Nightmode Dialogue downmix. However, music levels are much closer to source than the Nightmode Dialogue downmix, and due to inclusion of the LFE track, increasing the volume of the output while using this downmix formula can create an output stream that sounds truer to the 5.1 source than all other formulae I tested.

If you have the ability, I would highly recommend encoding your audio stream(s) using both this downmix formula and the Nightmode Dialogue downmix, and carefully comparing the waveforms of the two to determine which one is better.

The Nightmode Dialogue formula, created by Robert Collier on the Doom9 forum and sourced by Shane Harrelson in his answer, results in a far better downmix than the -ac 2 switch - instead of overly quiet dialogues, it brings them back to levels that are much closer to the source.

After converting many DTS movie tracks from 5.1 to 2.0 using eac3to, I have found the default eac3to channel mappings to result in very quiet dialogues and overly loud music and action scenes. Although the eac3to channel downmix coefficients have a scientific basis, they often do not sound good in practice bceause of low dialogue volume. This preset is for those looking for clear dialogues with left and right channel music still being audible but more in the background.

As you can see - front center (dialogues) come in properly now and stay at the original level - while the music and explosions remain a background effect and don't overpower you. This preset solves the problem of you having to constantly fiddle with the volume knob when watching DTS 5.1 converted to 2.0 movies in order to hear dialogues. (Especially for watching movies in the night where you don't want to wake others but still want to be able to hear dialogues).

Unfortunately, the music of this downmix formula is much lower than in the 5.1 source (which was likely by design considering Collier's intention to create a "nightmode" mix) and due to complete loss of the LFE track, the overall output audio doesn't sound as full or close to source as Dave_750's formula with boosted volume.

However, if for some reason you want to avoid boosting the overall volume of the stream, then the Nightmode Dialogue would likely be your best option - though again, I would highly recommend encoding your audio stream to both and comparing the waveforms of the two carefully.

This answer simply puts the Nightmode Dialogue downmix formula from Shane Harrelson's answer into a command to convert the audio stream in an MKV container. While the command given in this answer would work fine on such an audio stream, adapting it for a standalone audio track would give the error:

This is because the audio codec cannot be copied when downmixing - like all other changes FFmpeg makes to an output stream, a downmix requires that the track be re-encoded for the changes to be applied.

I found the answer Shane provided to provide too little of the other channels and too much of the center. Movies with headphones sounded off balance, with all dialogue and not enough background music/effects.

clev and slev should be .707, according to tables 5.9 and 5.10 in the aforementioned document, assuming a center/surround mix level of 0. Other values are provided in those tables which reduces the amount of center mix, which I don't find useful.

Implementations MAY use the matrices in Figures 4 through 9 toimplement downmixing from multichannel files using channel mappingfamily 1 (Section 5.1.1.2), which are known to give acceptableresults for stereo. Matrices for 3 and 4 channels are normalized soeach coefficient row sums to 1 to avoid clipping. For 5 or morechannels, they are normalized to 2 as a compromise between clippingand dynamic range reduction.

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