Re: Format Factory 32 Bits Full
Hercules Montero
Interface is minimal with a list of conversion types in left-hand column preview of selected format in right-hand column. Preview of selected size can be quickly changed from one format to another by simply clicking on desired conversion type. In lower-left corner of screen there is a text box that allows you to choose data that will be converted to a new format. Format Factory free to download full version has a very easy-to-use interface, with a menu on left side of screen and a preview window on right. Selecting a multimedia file from left menu brings it up for preview in preview window. This preview window allows user to preview video, as well as change some of settings of it. Software download Format Factory for PC has a box on bottom of screen with some simple instructions for how to use program. Interface of soft is simple and intuitive. It consists of a list of sizes which user can convert and a preview window. Available formats can be changed by clicking "Format" button in lower left corner. To convert files, simply click Select files button and select files. Files will be converted according to settings in Output settings window. You can choose to extract audio file from video file. To remove audio from video file, click Select video button and select video.
Application Format Factory app download is very easy to use. Only thing user needs to do is select multimdedia file from left menu, preview it, then click convert. This opens up conversion window, where user can change settings of video or audio file. There are three other menu options at top of screen, including one for user to preview video.
Product Format Factory Mac allows user to extract audio from video files, convert multimedia files from one size to another, export only audio from video, more. Software free Format Factory allows user to adjust settings of video, such as bit rate, frame rate, more. Application is a one-screen application with a simple interface. It has a menu with following functions: "Add Files", "Settings", "Profile", "Remove Format", "Remove Folder", "Remove File", "Preview", "Exit". Find more info on the format-factory.zendesk.com website.
Software Format Factory 64 bit is a free, open-source application that has a user-friendly interface and is easy to use. It is a one-stop-shop for all your media conversion needs. Application is free to download and use. Product has online documentation available.
Overall, install Format Factory is a powerful and convenient multimedia conversion tool with a simple and intuitive interface. It has a wide range of features, wide-range of support available. It was created with intention to help people who want to easily and quickly convert one image or multimedia format to another. This application allows to resize images, extract audio from multimedia and create animated GIFs from video clips.
I am not talking about the output having better quality than the input: obviously, that is not possible. (Except for going from a lossless format to the original wave.) I am talking whether an output with a higher bitrate than the input will have a better quality than it might otherwise have.
please consider that I am aware that converting between lossy formats is not recommendable. Only that in some cases an original cd/wave may be unavailable. The question is just about the usefulness of optionally increasing the bitrate when converting.
the most voted two answers below (this and this) seem to say different things, namely, the later says that Bitrates are not directly comparable and if the original audio is in a more efficient format, then the output (less efficient) audio should have a somewhat superior bitrate (the same idea here and here) - but while the less efficient is mp3, I am not sure which exactly are the more efficient formats. (is it aac?) (-- And in general the answers seem to fall in one of the two positions represented by the most voted answers.)
Of course you will never get anything back that was lost in the first place. On the contrary, encoding as mp3 will reduce the quality further. Every lossy format uses other means to reduce the amount of data that is stored, by (simplified) throwing away "unneeded" parts of the data. Round trip through a bunch of different formats and there won't be much left ...
So if you want to stay as close a possible to the quality your file has now, you should chose a higher bitrate. 320kbps are probably wasted space, but for mp3 something in the order between 128 and 192 is needed to maintain - or at least come close to - the quality of a more efficient 95kbps file.
Think about it this way: when it was converted from the original media (let's say a CD) it was compressed to fit the "content" in a smaller "box", and by doing so an amount of data has been lost (you may want to read about lossy and lossless formats).If you subsequently increase the bitrate, you are just making the "box" bigger, but the "content" is always the same.
First it's correct that you don't get more information from up sampling. But combining up sampling with a low pass (or interpolation) filter will get you a smoother curve. Passing this to the stereo should result in less noise produced from the stereo trying to reproduce the noise given by the original low sampling rate.
The important factor here is that you know something your stereo doesn't. Your stereo does not know noise from signal. It thinks that what you feed it is what you want. But you know the difference. You know you don't want the shape of the original signal, but a smoother version. So you can up sample and make a smooth curve, before feeding it to your stereo.
If you must re-encode it, the best result you can achieve is the same quality by choosing a losless codec like FLAC or ALAC. Or even uncompressed formats like WAV.
As an extreme case, suppose you have an uncompressed raw file with 16 bits per sample, stereo, at the sampling rate of 22 kHz. That amounts to 700 kbps. You encode it to MP3, high quality, 22 kHz, and get around, say, 64 kbps.
Suppose now we're doing the reverse, and want to encode a 64 kbps MP3 stream as RAW. Does it make sense to raise the data rate? You bet it does. If you did not - actually if you did not raise the data rate enough, and only went up to 350 kbps - the RAW format would allow for only half the sampling frequency. Or maybe only 8 bit per sample. Or maybe mono instead of stereo.
Actually a bit more than proportionally, because the second encoder, when cascaded with the first decoder, will always introduce an additional quality loss (unless you're using two lossless formats) that has to be compensated (even if you can't compensate all of it).
When the transcoding goes towards a higher compression for the same quality, then increasing the data rate does not make sense (actually it might well be that you're transcoding because the target format allows a better compression, and therefore the same quality with lower data rate).
My golden rule, however, is that information can only be destroyed - so transcode as little as possible, and always try to get as "near" as possible to the original source (in terms of transcoding "hops"). This will also achieve better compression and/or lower data rates, since you're not carrying aboard the noise and artifacts that the encoding process is heir to.
This is a complementary answer made to record what I consider to be the meaning of the other answers so far. There are different ideas floating here, maybe because my question was too general or vague. I have edited it to clarify, but the bad is done.
When the input is a video, the best way would be just to extract the audio file (for example, as specified here, for Linux, or with a program like the one mentioned here, for Windows, called SUPER. (After installing and taking care to avoid a bunch of adware that is proposed: Select the Output Process called "DeMux Extract Streams", after checking the second case in the upper corner of the window. Drag & Drop the file(s) you want to process. Click on "DeMux (Active Files)"). - Usually videos that may be the object of a such operation contain mp3 or aac audio.
If you are being forced to change formats, and convert between lossy formats, this most probably happens because you need mp3 files; also, there are cases where the audio of the input video is not an mp3. So, for a video, if it's not mp3, it will be in most cases an aac file. In this situation the bitrate of the mp3 output should be higher (in order to compensate the more inefficient bitrate of the mp3): for a 95kbps aac, the resulting mp3 should have a bitrate of about 128-192 kbps, etc.
There are some excellent technical descriptions of why this is a bad idea in this thread; to offer a different perspective, imagine that every time you make a lossy-compressed audio file (MP3, OGG, AAC), it's like dubbing a cassette tape. Even if you buy the most robust, highest-quality tape you can buy, every time you dub it, all that does is minimize the damage - it's still going to get a little more distorted. When it gets copied, you're always losing a little bit of quality you can never get back. It will never, ever, get "better".
For instance if you use the ffmpeg command line tool, you can give it the argument -acodec copy to instruct it to just copy audio data from one container to another without decoding and re-encoding it.
Overall, I would probably pick AAC, because it is widely supported, supports a wide range of bitrates and usually beats competitors at any bitrate. Furthermore, AAC has a low bitrate mode called HE-AAC which employs some sophisticated algorithms to reproduce high frequencies and stereo in a very bandwidth-preserving way.
Because of that, HE-AAC allows you to go as low as 32 kbps for music and 16 kbps for speech, while maintaining an acceptable listening experience. The European Broadcast Union has released a review of the different codecs:
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