waveform | simultaneously transcode and generate visuals | Audio Utils library

 by   andrewrk C Version: Current License: MIT

kandi X-RAY | waveform Summary

kandi X-RAY | waveform Summary

waveform is a C library typically used in Audio, Audio Utils applications. waveform has no bugs, it has no vulnerabilities, it has a Permissive License and it has low support. You can download it from GitHub.

Input: any format audio or video file.
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            kandi-support Support

              waveform has a low active ecosystem.
              It has 236 star(s) with 28 fork(s). There are 27 watchers for this library.
              OutlinedDot
              It had no major release in the last 6 months.
              There are 2 open issues and 18 have been closed. On average issues are closed in 48 days. There are no pull requests.
              It has a neutral sentiment in the developer community.
              The latest version of waveform is current.

            kandi-Quality Quality

              waveform has no bugs reported.

            kandi-Security Security

              waveform has no vulnerabilities reported, and its dependent libraries have no vulnerabilities reported.

            kandi-License License

              waveform is licensed under the MIT License. This license is Permissive.
              Permissive licenses have the least restrictions, and you can use them in most projects.

            kandi-Reuse Reuse

              waveform releases are not available. You will need to build from source code and install.
              Installation instructions, examples and code snippets are available.

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            waveform Key Features

            No Key Features are available at this moment for waveform.

            waveform Examples and Code Snippets

            Inverse Fourier Transform .
            pythondot img1Lines of Code : 118dot img1License : Non-SPDX (Apache License 2.0)
            copy iconCopy
            def inverse_stft(stfts,
                             frame_length,
                             frame_step,
                             fft_length=None,
                             window_fn=window_ops.hann_window,
                             name=None):
              """Computes the inverse [Short-time Fourier Transf  
            Inverse of the inverse mdct .
            pythondot img2Lines of Code : 81dot img2License : Non-SPDX (Apache License 2.0)
            copy iconCopy
            def inverse_mdct(mdcts,
                             window_fn=window_ops.vorbis_window,
                             norm=None,
                             name=None):
              """Computes the inverse modified DCT of `mdcts`.
            
              To reconstruct an original waveform, the same window function   
            Returns a function that computes the inverse window function for a given frame step .
            pythondot img3Lines of Code : 56dot img3License : Non-SPDX (Apache License 2.0)
            copy iconCopy
            def inverse_stft_window_fn(frame_step,
                                       forward_window_fn=window_ops.hann_window,
                                       name=None):
              """Generates a window function that can be used in `inverse_stft`.
            
              Constructs a window that is equ  

            Community Discussions

            QUESTION

            I'm getting an syntax error in my VHDL code near counter
            Asked 2021-Jun-10 at 13:14

            I'm trying to simulate a pulse width modulate (PMW) waveform generator and getting a syntax error in ISE. Checked fuse.xmsgs and found out it's near counter. Can someone point out the syntax error, please?

            ...

            ANSWER

            Answered 2021-Jun-10 at 13:14

            counter count := 0;

            This is illegal syntax, as you didnt declare the object class (signal, constant, variable). You need to use the format:

            signal count : counter := 0

            This is also illegal, as you are comparing an integer to a std_logic_vector that you havent included a package for. You need to convert the slv to an unsigned

            if (count <= unsigned(width)) then

            And finally, reset is missing from the sensitivity list

            Source https://stackoverflow.com/questions/67921622

            QUESTION

            Multiple Core Usage/Parallelize Map Function for an RQA
            Asked 2021-Jun-10 at 09:47

            I am currently trying to run a windowed CRQA in R using heart and respiration waveform signals. I have to run 94 windowed CRQA, where each signal has 20000 - 50000 data points. Hence the computational load is relatively high and takes forever. Therefore I am trying to get R to increase the memory size and use multiple cores. Both this does not seem to work with the folllowing code:

            ...

            ANSWER

            Answered 2021-Jun-10 at 09:47

            You need to use clusterMap() or another function from the parallel package rather than base::Map.

            Look those options up with: ?parallel::clusterMap.

            You are also currently mixing things up by introducing registerDoParallel(cl), which is from the foreach package. That would require you to then use foreach() and its helper %dopar%. If you do not use foreach() then you do not need registerDoParallel(cl).

            The relevant part of your code would look something like this with clusterMap(). I've tidied it a bit but I can't test it on my machine.

            Source https://stackoverflow.com/questions/67916760

            QUESTION

            Animate CAShapelayer content when new item is added
            Asked 2021-May-29 at 12:59

            I need to render moving audio waveform like iOS voice memo app. Here I maintain waveform:[Int] rms amplitude of waves. Now when new waves come add it in waveform[Int] and I add new UIBezierPath line at right of CAShapeLayer and translate whole CAShapeLayer by 5 points.

            But translation animation is not so smooth. Could you please suggest any better approach for it?

            My current implementation:

            ...

            ANSWER

            Answered 2021-May-29 at 12:59

            Don't try to do your animation in draw(rect:). That puts all the work on the CPU, not the GPU, and does not take advantage of the hardware-accelerated animation in iOS.

            I would suggest instead using a CAShapeLayer and a CABasicAnimation to animate your path.

            Install the CGPath from your UIBezierPath into the CAShapeLayer, and then create a CABasicAnimation that changes the path property of the shape layer.

            The trick to getting smooth shape animations is to have the same number of control points for every step in the animation. Thus you should not add more and more points to your path, but rather create a new path that contains a graph of the last n points of your waveform.

            I would suggest keeping a ring buffer of the n points you want to graph, and building a GCPath/UIBezierPath out of that ring buffer. As you add more points, the older points would "age out" of the ring buffer and you'd always graph the same number of points.

            Edit:

            Ok, you need something simpler than a ring buffer: Let's call it a lastNElementsBuffer. It should let you add items, discarding the oldest element, and then always return the most recent elements added.

            Here is a simple implementation:

            Source https://stackoverflow.com/questions/67660015

            QUESTION

            abs() returns the same output for different FFT inputs
            Asked 2021-May-26 at 23:03

            I have a 1024 samples and I chucked it into 32 chunks in order to perform FFT on it, below is the output from FFT:

            ...

            ANSWER

            Answered 2021-May-26 at 22:27

            Your signal is a sine wave. You chop it up. Each segment will have the same frequency components, just a different phase (shift). The FFT gives you both the magnitude and phase for each frequency component, but after abs only the magnitude remains. These magnitudes are necessarily the same for all your chunks.

            Source https://stackoverflow.com/questions/67713216

            QUESTION

            how to get buffer from AVAudioEngine's installTap at high frequency
            Asked 2021-May-26 at 13:13

            I am looking to render audio waveform on screen like iOS voice memo app, so I am using AVAudioEngine and installed Tap on input node. But it gives fastest call back on 0.1sec frequency , I need to get buffer data on faster frequency , so that can draw more and more waveform on screen for smooth and realtime visualisation.

            Could you please sugger better approach for it ?

            let inputNode = self.audioEngine!.inputNode let format = inputNode.inputFormat(forBus: 0)

            ...

            ANSWER

            Answered 2021-May-26 at 13:13

            I think you misunderstand how to handle this. Instead of getting very small amounts of new data at a high frequency, you should factor your code to get a larger chunk of data on a longer time interval.

            You can then set up your display code to start an animation to animate in a graph of the new data over the time interval you are given. 1/10th of a second is actually a pretty short time interval for an animation.

            I would suggest averaging the amplitude of the data for that 1/10th of a second time interval and animating that new average data into place.

            BTW, see the update I posted to the answer to your question about creating an animated graph of the data.

            The animation I came up with looks like this:

            Source https://stackoverflow.com/questions/67661093

            QUESTION

            assign not updating result value in testbench in EDA playground
            Asked 2021-May-23 at 14:27

            I am using EDA Playground with Aldec Riviera simulator, and I have this module here:

            ...

            ANSWER

            Answered 2021-May-23 at 14:27

            In the testbench, you declared the result signal, but it is not connected to anything. You probably intended it to be driven by the alu output of the same name. In that case, you should connect it to the instance:

            Change:

            Source https://stackoverflow.com/questions/67660548

            QUESTION

            OpenGL draws outside window bounds
            Asked 2021-May-21 at 11:58

            while drawing a GL_LINE_STRIP I noticed the drawing is outside of window bounds, I can't see half of the drawing.

            I want to center it within the window and automatically resize it to fit it. can someone redirect me to how can I achieve that?

            this is what I'm seeing:

            I want it to be in that size & position:

            This is my code:

            ...

            ANSWER

            Answered 2021-May-21 at 11:06

            To solve the isseu, you need to know the bound of your curve. You have to change the the orthographic projection:

            glOrtho(-6, 6, -6, 6, -1, 1);

            In Orthographic Projection, the view space coordinates are linearly mapped to the clip space coordinates. The viewing volume is defined by 6 distances (left, right, bottom, top, near, far). The values for left, right, bottom, top, near and far define a cuboid (box).

            If you want to center the view on the curve, the center of the projection must be the center of the curve. e.g.:

            Source https://stackoverflow.com/questions/67635049

            QUESTION

            Interpolating frequencies
            Asked 2021-May-20 at 15:57

            This is more of a math problem than a programming problem, but it has now come up for me twice in two different programming projects.

            I would like to create a function, w_sweep(t, f0, t0, f1, t1), that samples at time t a sine wave that sweeps through frequencies from f0 when t == t0 to f1 when t == t1. (You can assume frequencies are in Hz and times are in seconds. I don't care what happens when t is outside the range t0..t1.)

            I started by writing a simpler function that uses a constant frequency:

            ...

            ANSWER

            Answered 2021-May-20 at 14:29

            It can be wise to look at the reverse, the change in periods. For a frequency f0, you have a period T0 = 1/f0. For a frequency f1, you have a period T1=1/f1.

            Now, when you evaluate w_sweep(t), you're not using one period T0, then a period lerp(T0,T1,...) etcetera. Instead, you determine a single period T = 1.0/std::lerp(f0, f1, (t - t0)/(t1 - t0)), and then calculate sin(2*pi*t/T) which is of course sin(2*pi*frac(t/T)).

            I think what you want (not entirely sure) is f=1/lerp(1/f0, 1/f1, (t - t0)/(t1 - t0))

            Source https://stackoverflow.com/questions/67621677

            QUESTION

            May I know why EQ=1, but the output no response?
            Asked 2021-May-20 at 01:36

            I am trying to build a Finite State Machine vending machine which consists of Datapath Unit and Control Unit. The attached links is the Control unit which consists the input of EQ(Equal), GT(Greater) and product. When product is "1" and either EQ or GT is "1", the output will be out=product. However, in my problem, the verilog code shows correct for GT but not EQ. It seem that the output cannot response to EQ when it is high.

            My design of the state diagram. State Diagram

            My Verilog code. Verilog code

            The result. Result Waveform

            ...

            ANSWER

            Answered 2021-May-19 at 16:00

            The answer is very simple. Add ps and ns to your simulation charts and you will understand why.

            At the beginning of your simulation you are in state S0. When the product is bigger than 0 (first yellow mark) you go to state S1. Then you are waiting for EQ or GT, but EQ fall down one clock cycle ago so next GT arrive.

            Set EQ and GT one clock cycle later.

            Source https://stackoverflow.com/questions/67596390

            QUESTION

            Java audio... getting from byte[] of raw waveform to playable Clip
            Asked 2021-May-18 at 23:23

            I'm trying to figure out how to play a raw 8-bit (signed) waveform using Java, and struggling because it seems like every API example I find assumes media playback from a file. Here's what I have so far (shortened for clarity):

            ...

            ANSWER

            Answered 2021-May-18 at 23:23

            Call the constructor directly like this:

            Source https://stackoverflow.com/questions/67594697

            Community Discussions, Code Snippets contain sources that include Stack Exchange Network

            Vulnerabilities

            No vulnerabilities reported

            Install waveform

            Install libgroove dev package. Only the main library is needed. Install libpng and zlib dev packages.
            Install libgroove dev package. Only the main library is needed.
            Install libpng and zlib dev packages.
            make

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