How To Drastically Reduce Your Input Delay In F...
How To Drastically Reduce Your Input Delay In F...
One important detail of this counter is that it reports the maximum user input delay within a configurable interval. This is the longest time it takes for an input to reach the application, which can impact the speed of important and visible actions like typing.
For example, in the following table, the user input delay would be reported as 1,000 ms within this interval. The counter reports the slowest user input delay in the interval because the user's perception of "slow" is determined by the slowest input time (the maximum) they experience, not the average speed of all total inputs.
Next, let's look at the User Input Delay per Session. There are instances for each session ID, and their counters show the user input delay of any process within the specified session. In addition, there are two instances called "Max" (the maximum user input delay across all sessions) and "Average" (the average acorss all sessions).
An important thing to remember when using this performance counter is that it reports user input delay on an interval of 1,000 ms by default. If you set the performance counter sample interval property (as shown in the following screenshot) to anything different, the reported value will be incorrect.
I found three commands that helped me reduce the delay of live streams. The first command its very basic and straight forward, the second one it's been combined with other options which might work differently on each environment and the last command it is a hacky version that I found in the documentation It was useful at the beginning but now the first option is more stable.
Consider using the filter option -vf setpts=0. This makes all frames display as soon as possible without adding any delay for the framerate. This will allow the stream to catch up in case it falls behind, which I've found to happen if I move or resize the ffplay window. However, this could make the video look choppy if your video data is being received at an inconsistent rate.
The constants Fsin, Fsout and D in the equations above are the input sample rate, output sample rate, and the resampling output delay respectively. For many multirate filtering applications, it is useful to find Fsout and D for a given input sample rate Fsin.
You can use the outputDelay function to calculate the resampling output delay D and output sample rate FsOut for a given filter object operating at rate FsIn. This function is available for any DSP System object that supports filter analysis methods. For a list of supported objects, refer to the outputDelay. The returned delay value D is specified in the natural units of the interpolated signal (usually seconds), corresponding to the input sample rate.
To eliminate the high frequency noise, design a lowpass filter and apply it to the signal. This lowpass has a cutoff at 15% of the Nyquist frequency with a transition width of 10%. Plot the input against the filtered output on the same graph. Note the delay between the input and the filtered signal.
To align the input with the output, shift the output back in time by D units, of shift the input forward in time by the same amount. You can perform such a shift using the TimeDisplayOffset property of the timescope object. When you specify a vector in TimeDisplayOffset instead of a scalar, each input channel of the timescope object has its own delay, corresponding to the entries of the vector in TimeDisplayOffset. Set the first channel (input) delay to D, and keep the second channel (output) with no delay. The two channels are now synchronized.
You can specify or override the input sample rate by using the named argument FsIn. For example, calculate the output delay assuming that the input sample rate is 2 kHz instead of 10 Hz. Note that the returned delay value changes accordingly to reflect the new time units.
In any case, since the group delay is not constant, you need to specify the frequency from which the group delay is sampled. The outputDelay function accepts this frequency (specified in input sample rate units) through the parameter Fc.
Some filter designs have a nonlinear phase, yet still have a relatively flat group delay on subbands. For example, any FIR designed using the designFracDelayFIR function has a relatively flat group delay. Other examples include the dsp.IIRHalfbandDecimator and the dsp.FIRHalfbandInterpolator filters that are operating in the Quasi-linear phase design mode. The Quasi-linear design mode compensates for the nonlinear phase of the IIR on the passband and considerably reduces the distortion.
For a filter with nonlinear phase stages, changing Fc alters the output delay D. The outputDelay function can calculate the interval of input frequencies B = [f1, f2] around Fc that have a delay value close to D up to a tolerance, or D(f)-D(Fc)
The outputDelay function can be used even if the group delay is not flat, given that the input signal is a narrowband signal. The band measurement returned from outputDelay can be used to determine the maximal bandwidth for the signal subject to a delay tolerance.
Filter a modulated signal through the filter, and plot the input against the output with the apropriate delay. As expected, the delayed input appears synchronized with the output under the same envelope (the delayed baseband signal), but have a slight phase shift - the phase delay of the carrier signal.
For band measurement, the cascade must be reducible to a single filter stage using noble identities. For example, the cascade g5 can be reduced to have a single convolution stage. Measure the input band for that filter.
Designed for vocals, the Echo Master is a lo-fi, analog tape-style delay unit that can transform your voice into a chaotic realm of noise and feedback. The effects loop allows additional effects to be placed in the signal chain, making it easy to use with any of your favorite effects pedals on your vocal sound without any adapters. 041b061a72