ffmpeg -ss 10 -t 6 -i input.mp3 output.mp3 says -t is not an input option, keeping it for the next output; consider fixing your command line. and gives me a file that's got 8 seconds of audio starting from 10s and then some 9 or 10 seconds of silence.

I used Quicktime and it has silent audio at the end. The description of the output file in finder says like 14 seconds. When I use VLC, it plays for the correct 6 seconds and stops, even though its duration in the file browser in VLC says 14. My MPlayer doesn't work properly. I also did the preview audio in Finder, and it plays the 6 seconds properly and then stops. But the round seeker bar of the MP3 didn't reach the end. And it also says 14 seconds instead of 6.


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Time stretching is the process of changing the speed or duration of an audio signal without affecting its pitch. Pitch scaling is the opposite: the process of changing the pitch without affecting the speed. Pitch shift is pitch scaling implemented in an effects unit and intended for live performance. Pitch control is a simpler process which affects pitch and speed simultaneously by slowing down or speeding up a recording.

These processes are often used to match the pitches and tempos of two pre-recorded clips for mixing when the clips cannot be reperformed or resampled. Time stretching is often used to adjust radio commercials[1] and the audio of television advertisements[2] to fit exactly into the 30 or 60 seconds available. It can be used to conform longer material to a designated time slot, such as a 1-hour broadcast.

The simplest way to change the duration or pitch of an audio recording is to change the playback speed. For a digital audio recording, this can be accomplished through sample rate conversion. When using this method, the frequencies in the recording are always scaled at the same ratio as the speed, transposing its perceived pitch up or down in the process. Slowing down the recording to increase duration also lowers the pitch, while speeding it up for a shorter duration respectively raises the pitch, creating the so-called Chipmunk effect. When resampling audio to a notably lower pitch, it may be preferred that the source audio is of a higher sample rate, as slowing down the playback rate will reproduce an audio signal of a lower resolution, and therefore reduce the perceived clarity of the sound. On the contrary, when resampling audio to a notably higher pitch, it may be preferred to incorporate an interpolation filter, as frequencies that surpass the Nyquist frequency (determined by the sampling rate of the audio reproduction software or device) will create usually undesired sound distortions, a phenomenon that is also known as aliasing.

The phase vocoder technique can also be used to perform pitch shifting, chorusing, timbre manipulation, harmonizing, and other unusual modifications, all of which can be changed as a function of time.

Another method for time stretching relies on a spectral model of the signal. In this method, peaks are identified in frames using the STFT of the signal, and sinusoidal "tracks" are created by connecting peaks in adjacent frames. The tracks are then re-synthesized at a new time scale. This method can yield good results on both polyphonic and percussive material, especially when the signal is separated into sub-bands. However, this method is more computationally demanding than other methods.[citation needed]

Rabiner and Schafer in 1978 put forth an alternate solution that works in the time domain: attempt to find the period (or equivalently the fundamental frequency) of a given section of the wave using some pitch detection algorithm (commonly the peak of the signal's autocorrelation, or sometimes cepstral processing), and crossfade one period into another.

This is called time-domain harmonic scaling[5] or the synchronized overlap-add method (SOLA) and performs somewhat faster than the phase vocoder on slower machines but fails when the autocorrelation mis-estimates the period of a signal with complicated harmonics (such as orchestral pieces).

This is much more limited in scope than the phase vocoder based processing, but can be made much less processor intensive, for real-time applications. It provides the most coherent results[citation needed] for single-pitched sounds like voice or musically monophonic instrument recordings.

High-end commercial audio processing packages either combine the two techniques (for example by separating the signal into sinusoid and transient waveforms), or use other techniques based on the wavelet transform, or artificial neural network processing[citation needed], producing the highest-quality time stretching.

These techniques can also be used to transpose an audio sample while holding speed or duration constant. This may be accomplished by time stretching and then resampling back to the original length. Alternatively, the frequency of the sinusoids in a sinusoidal model may be altered directly, and the signal reconstructed at the appropriate time scale.

Pitch-corrected audio timestretch is found in every modern web browser as part of the HTML standard for media playback.[10] Similar controls are ubiquitous in media applications and frameworks such as GStreamer and Unity.

I shot an interview with a Canon C100 back in December. I had two channels of audio going into the camera. Boom and Lav. When I brought it into Premiere, I noticed an echo in the timeline. It sounds as if the audio is off by a few subframes. I did some digging and found out that you can switch the sequence to show audio time units. From there I was able to slip one channel of audio over at the subframe level to eliminate the echo.

I do the same thing as Thegent - Guitar to Reaper, then video on my iphone. I pull both recordings into Davinci Resolve and let the software sync the audio tracks. Then I just disable the included audio from the video track.

For a little while, I was piping the audio from the DAW out to the video capture software, but it was kind of a hassle. Using a console is a bit heavy handed, but you just turn it on and go, which is nice.

There's no MS-documented time limit, but in practice, users do find that some versions of PPT do truncate audio. As it happens, I've been looking into this a bit and have been collecting reports from users who run into the problem. So far 40 seconds seems to be the magic number. Narration shorter than that doesn't seem to get cut off. It appears to be more a bug in how PPT "reads" the length of the audio file than an actual limit, so there's no fix until MS comes up with one. I suspect you could work around the problem by keeping your narrations down to say 35 or 30 seconds per slide; for slides that require longer narration, add a second duplicate slide to add the remainder of the narration to and then set the transition to the duplicate slide to automatic.

I have PPT version 2111. I not only find that it truncates my audio, but it truncates my entire presentation. It won't let me record past slide 26 of a 60 slide deck. The current file size is just under 32 GB. And PPT suggests that what is recorded is just over 18 minutes.

I tried to mute the scrubbing sound in the function before seeking the time (from your script). This worked only sort of. When the volume of the audio was down or the audio was muted, it would indeed get rid of the scrubbed audio sound, but when I tried to add the audio back after the time was seeked, the audio would sound garbled on the scrub as though it had never been muted.

Successfully mute the audio (or turn the volume down) while the drag action is scrubbing (whether the timeline/audio is currently playing or not), and then unmute the audio (or turn the audio back up) once the drag action is done?

What you can do is listen for the start and end on the drag (Hype has the ability to listen for this event built in) and then pause and play the sounds plus timeline when these events fire. Change the code I gave you above to

I added your code and that took care of the audio scrub sound issue wonderfully!

When the timeline IS playing and I try to control the timeline On Drag, it works absolutely perfectly! (I appreciate your tidying the code as well, thanks for the refactoring example!).

Perhaps would it work to make a global variable that can be set to something like isPlaying or notPlaying when the timeline is manipulated (started, paused, or continued) outside of the function? That way, the global variable could be checked and reset inside the function rather than checking for hypeDocument.isPlayingTimelineNamed(...).

In the Project panel, clips that contain both video and audio appear as a single item, represented by . When you add the clip to the sequence, however, the video and audio appear as two objects, each in its appropriate track (provided you specified both the video and audio sources when adding the clip).

The video and audio portions of the clip are linked so that when you drag the video portion in a Timeline panel, the linked audio moves with it, and conversely. For this reason, the audio/video pair is called a linked clip.

Ordinarily, all editing functions act on both parts of a linked clip. When you want to work with the audio and video individually, you can unlink them. When you do, you can use the video and audio as though they were not linked. Even so, Premiere Pro tracks the link. If you relink the clips, they indicate whether they have been moved out of sync, and by how much. You can have Premiere Pro automatically resynchronize the clips.

Right-click (Windows) or Control-click (macOS) the number that appears at the In point in a Timeline panel of the out-of-sync video or audio clip. The number indicates the amount of time the clip is out of sync with its accompanying video or audio clip.

"Be aware that: On a compressed audio track position does not necessary reflect the actual time in the track Compressed audio is represented as a set of so-called packets. The length of a packet depends on the compression settings and can quite often be 2-3 seconds per packet. See Also: timeSamples variable." 2351a5e196

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