Convert 44.1 Khz Audio To 48khz Premiere

May 06, 2016 sr-convert is most often used to convert between 44.1 kHz and 48 kHz. For example, if you record your own music at 48 kHz, you can downconvert it to 44.1 kHz for burning to CD. Or, if you rip CDs, you can upconvert them to 48 kHz before you ogg-encode them, which results in slightly better sound quality.

  1. Convert 44.1 Khz Audio To 48khz Premiere Software
  2. Convert 44.1 Khz Audio To 48khz Premiere Plus
(Redirected from Sample rate conversion)

Sample-rate conversion is the process of changing the sampling rate of a discrete signal to obtain a new discrete representation of the underlying continuous signal.[1] Application areas include image scaling[2] and audio/visual systems, where different sampling rates may be used for engineering, economic, or historical reasons.

  1. Convert MP3 file. Using Zamzar it is possible to convert MP3 files to a variety of other formats. Mp3 to aac (Advanced Audio Coding File) mp3 to ac3 (AC3 Audio File) mp3 to flac (Free Lossless Audio Codec) mp3 to m4a (Compressed video file) mp3 to m4r (iPhone Ringtone File) mp3 to mp4 (MPEG-4 Video File).
  2. Question to audio experts 48 conversion to 44.1 kHz: I'm recording my new album and want to use the Axe Fx XL as my main guitar sound. All the other musicians already recorded using 44.1kHz. I know that the AxeFx works on 48 kHz on Logic. Do you guys experience any loss in quality when.

For example, Compact Disc Digital Audio and Digital Audio Tape systems use different sampling rates, and American television, European television, and movies all use different frame rates. Sample-rate conversion prevents changes in speed and pitch that would otherwise occur when transferring recorded material between such systems.

Convert 44.1 Khz Audio To 48khz Premiere

Within specific domains or for specific conversions, the following alternative terms for sample-rate conversion are also used: sampling-frequency conversion, resampling, upsampling, downsampling, interpolation, decimation, upscaling, downscaling. The term multi-rate digital signal processing is sometimes used to refer to systems that incorporate sample-rate conversion.

Techniques[edit]

Conceptual approaches to sample-rate conversion include: converting to an analog continuous signal, then re-sampling at the new rate, or calculating the values of the new samples directly from the old samples. The latter approach is more satisfactory, since it introduces less noise and distortion.[3] Two possible implementation methods are as follows:

  1. If the ratio of the two sample rates is (or can be approximated by)[A][4] a fixed rational number L/M: generate an intermediate signal by inserting L − 1 zeros between each of the original samples. Low-pass filter this signal at half of the lower of the two rates. Select every M-th sample from the filtered output, to obtain the result.[5]
  2. Treat the samples as geometric points and create any needed new points by interpolation. Choosing an interpolation method is a trade-off between implementation complexity and conversion quality (according to application requirements). Commonly used are: ZOH (for film/video frames), cubic (for image processing) and windowedsinc function (for audio).

The two methods are mathematically identical: picking an interpolation function in the second scheme is equivalent to picking the impulse response of the filter in the first scheme. Linear interpolation is equivalent to a triangular impulse response; windowed sinc approximates a brick-wall filter (it approaches the desirable 'brick wall' filter as the number of points increase). The length of the impulse response of the filter in method 1 corresponds to the number of points used in interpolation in method 2.

In method 1, a slow pre-computation (such as the Remez algorithm) can be used to obtain an optimal (per application requirements) filter design. Method 2 will work in more general cases, e.g. where the ratio of sample rates is not rational, or two real-time streams must be accommodated, or the sample rates are time-varying.

See decimation and upsampling for further information on sample-rate conversion filter design/implementation.

Examples[edit]

Film and television[edit]

The slow-scan TV signals from the Apollo moon missions were converted to the conventional TV rates for the viewers at home. Digital interpolation schemes were not practical at that time, so analog conversion was used. This was based on a TV rate camera viewing a monitor displaying the Apollo slow-scan images.[6]

Convert 44.1 khz audio to 48khz premiere pro

Movies (shot at 24 frames per second) are converted to television (roughly 50 or 60 fields[B] per second). To convert a 24 frame/sec movie to 60 field/sec television, for example, alternate movie frames are shown 2 and 3 times, respectively. For 50 Hz systems such as PAL each frame is shown twice. Since 50 is not exactly 2×24, the movie will run 50/48 = 4% faster, and the audio pitch will be 4% higher, an effect known as PAL speed-up. This is often accepted for simplicity, but more complex methods are possible that preserve the running time and pitch. Every twelfth frame can be repeated 3 times rather than twice, or digital interpolation (see above) can be used in a video scaler.

Audio[edit]

Audio on Compact Disc has a sampling rate of 44.1 kHz; to transfer it to a digital medium that uses 48 kHz, method 1 above can be used with L = 160, M = 147 (since 48000/44100 = 160/147).[5] For the reverse conversion, the values of L and M are swapped. Per above, in both cases, the low-pass filter should be set to 22.05 kHz.

See also[edit]

Sample rate conversion in multiple dimensions:

Techniques and processing that may involve sample-rate conversion:

Notes[edit]

Premiere
  1. ^For example, the irrational ratio 21/12, corresponding to one equal-temperament semitone, might be approximated by 196/185 (roughly 0.99994 semitones).
  2. ^A field is half of an interlaced frame – just the odd or even lines.

References[edit]

Convert 44.1 Khz Audio To 48khz Premiere Software

  1. ^Oppenheim, Alan V.; Schafer, Ronald W.; Buck, John R. (1999). Discrete-time signal processing (2nd ed.). Upper Saddle River, N.J.: Prentice Hall. ISBN0-13-754920-2. Also available at https://d1.amobbs.com/bbs_upload782111/files_24/ourdev_523225.pdf
  2. ^Lyons, Richard (2010). '10. Sample Rate Conversion'. Understanding Digital Signal Processing. Prentice Hall. ISBN978-0137027415. In satellite and medical image processing, sample rate conversion is necessary for image enhancement, scale change, and image rotation
  3. ^Antoniou, Andreas (2006). Digital Signal Processing. McGraw-Hill. p. 830. ISBN0-07-145424-1. An alternative and more satisfactory approach...
  4. ^Partch, Harry (2009). Genesis of a Music (Second ed.). Da Capo Press. p. 101. ISBN978-0306801068. The ratios of the 'equal semitone' are, with progressive accuracy: 18/17, 89/84, 196/185
  5. ^ abRajamani, K.; Yhean-Sen Lai; Furrow, C. W. (2000). 'An efficient algorithm for sample rate conversion from CD to DAT'(PDF). IEEE Signal Processing Letters. 7 (10): 288. doi:10.1109/97.870683.
  6. ^Bill Wood (2005). 'Apollo Television'(PDF). NASA. See the section 'Use of the RCA Slow Scan Converter'.

Further reading[edit]

  • Crochiere, R.E.; Rabiner, L.R. (1983). Multirate Digital Signal Processing. Englewood Cliffs, NJ: Prentice-Hall. ISBN0136051626.

Convert 44.1 Khz Audio To 48khz Premiere Plus

External links[edit]

Convert 44.1 khz audio to 48khz premiere pro
  • The Quest For The Perfect Resampler (PDF)
Retrieved from 'https://en.wikipedia.org/w/index.php?title=Sample-rate_conversion&oldid=1003832377'
I have been trying to bounce a 16 track Logic Audio song down to one single stereo .wav file. At first I was bouncing them down to a 24 bit .wav file, and it sounded fine in a Windows MediaPlayer type application but when I burned it to CD is had a robotic digital noise type crap added during the burning process. And then I finally got smart and bounced it down to 16 bit .wav. Then the noise was STILL there.
I record via the Delta-66 into my Logic Audio Platinum 5. And I record at a 24-bit resolution and a 48kHz sampling rate. Now, I know that CD audio is 16-bit/44.1kHz. In Logic Audio, when I go to the Track mixer to bounce down the selected tracks to one stereo .wav file, it asks which bit resolution I want to use (24 or 16 bit). However I do not recall ever seeing anything about converting the sampling rate to 44.1.
Does the software just assume that I want 44.1 and during the bouncedown it is converting it to 48kHz? How am I supposed to get a 24-bit/48kHz track down to 16/44.1?
And would dithering help me eliminate my robotic sounding digital noise?

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