Encoding wav files

Just to clarify, PCM is just the audio data uncompressed straight up. Like what you have on a CD. Don't need codecs of any kind for this. JaredBusch said in How can I find the current wav encoding :.

Today I've issues with internet None had answered when I've started posting That's exactly what prevents them from playing in a regular media player.

MS PCM is the standard wave file. RojoLoco said in How can I find the current wav encoding :. There might be some speeding up or slowing down if you don't do the right steps to change the sample rate can't remember the whole process in Audacity.

If you can email me a small sample, I can provide better instructions. RojoLoco I was trying to stay in the built in functions, but was running out of ideas.

Had this not worked, I would probably have had to go the Audacity route. Good thing you brought this up. I had to dig around before I could find that setting anywhere.

But it's under the recording quota. IMHO you should have a lot of recordings to enable it in the first place. In just 10GB you have room for hours of non-compressed recordings. Also, WAV is patented by Microsoft, which gives it the upper hand incompatibility. That would defeat the purpose of having them there in the first place. Depending on what you need out of your audio files, each type of audio file has its advantages and disadvantages. For the average listener, you usually would not focus on the file type of your songs.

However, knowing the pros and cons of each file format can help you decide which is best for you under your specific circumstances. WAV files offer the best sound quality. These audio files are also used most commonly by streaming services and have a higher compatibility rate than FLAC files.

WAV files are also great for sound engineers who need to edit audio files. However, WAV files are large, so you cannot fit as many files on one device. WAV files are also more difficult to share due to their large size, but this might be by design. In the case of sound quality, FLAC files are typically just as great.

If there is a difference in sound quality, it is usually negligible. FLAC files are also easily sharable and can even be found online for free since they are open-source file formats.

On the other hand, not every device or streaming software is compatible with FLAC files. This is a major hindrance if you cannot listen to your collection of audio files.

Whether one is better than the other depends on what conveniences you are hoping to find. If you want to add a huge amount of songs to your playlist, it may be wise to go with FLAC, but if you want consistent sound quality and compatibility, WAV is the best choice. Save my name, email, and website in this browser for the next time I comment. SaveNetRadio is supported by readers. To make it very simple the math is definitely above me one bit gives you about 6dB of SNR.

It is extremely rare for normal people to actually need more for a finished "mastered" result. Basically no sound equipment can playback even close to that. But we tend to use 24 bit for a bit of extra safety. I tend to record 24bit and aim for dB full scale as max and add a bit of gain before export. Works for me. So from that point of view I would say that it is perfectly good to export 48kHz at 24bits as.

No sound will be lost, no extra noise is added. And most if not all programs can import that format. But here comes the caveat: the format cannot give you anything above 0dB simplified in the sound wave.

So if the sound ever goes outside of this it will distort. It is similar to if you record from a microphone and sets gain to high. One way to handle that is normalize the sound: just about every sound program has a normalize function. Set it to -1 dB or so in order to be a little on the safe side and export the result.

One alternative is to export as 32 bit float. The float format allows a signal a lot stronger than 0dB, several hundred dB-s. This means that on your export the sound will not be distorted. Not all programs can receive 32 bit float, so you need to check what the receiving end expects.

If you ask me the 64 bit float as export is simply stupid. More than half the file will contain no useful information at all. There are however more to the art of "mastering". One part is to investigate closer what the receiving end does with the sound file.

Often the receiving and applies various types of processing, and if you want your sound to be the best you might want to tailor the export to that. If you export to say Youtube, they will do something like normalizing. They will pull down all of the sound in the file to make the single max volume part below some value my guess is dB. In order to make your listener happy this means that you should not export a sound file with large differences: a classical mistake is when the intro music is very strong followed by extremely weak spoken word.

In some other media, say on radio, there is often a compressor applied to the sound.