FFTConvolver | Audio convolution algorithm in C++ for real time audio | Audio Utils library

 by   HiFi-LoFi C++ Version: Current License: MIT

kandi X-RAY | FFTConvolver Summary

kandi X-RAY | FFTConvolver Summary

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

FFTConvolver is a C++ library for highly efficient convolution of audio data (e.g. for usage in real-time convolution reverbs etc.).
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              FFTConvolver has a low active ecosystem.
              It has 246 star(s) with 49 fork(s). There are 17 watchers for this library.
              OutlinedDot
              It had no major release in the last 6 months.
              There are 3 open issues and 3 have been closed. On average issues are closed in 306 days. There are 2 open pull requests and 0 closed requests.
              It has a neutral sentiment in the developer community.
              The latest version of FFTConvolver is current.

            kandi-Quality Quality

              FFTConvolver has no bugs reported.

            kandi-Security Security

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

            kandi-License License

              FFTConvolver 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

              FFTConvolver releases are not available. You will need to build from source code and install.

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            FFTConvolver Key Features

            No Key Features are available at this moment for FFTConvolver.

            FFTConvolver Examples and Code Snippets

            No Code Snippets are available at this moment for FFTConvolver.

            Community Discussions

            QUESTION

            Audio Convolution output is longer than input, how can i get around this when feeding data back to a stream of fixed length?
            Asked 2020-Oct-15 at 12:18

            After taking audio data from a stream of length x, the data is then convolved with an impulse response of length 256. This gives the output vector a length of (x + 256 - 1). When the data is then fed back into a stream of length x there are 255 samples of overshoot that then causes popping and clicking.

            Is there a work around for this? Im not 100% on how to merge the larger than original buffer into the output again without losing random samples or causing this issue.

            I left out the larger irrelevent parts of the code, it all works its just this issue i need fixed. Its just here to give an insight into the problem.

            Code:

            ...

            ANSWER

            Answered 2020-Oct-15 at 12:18

            You remember the 255 extra samples and add their values to the 255 samples at the beginning of the next output block.

            For example:

            Source https://stackoverflow.com/questions/64360217

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

            Vulnerabilities

            No vulnerabilities reported

            Install FFTConvolver

            You can download it from GitHub.

            Support

            For any new features, suggestions and bugs create an issue on GitHub. If you have any questions check and ask questions on community page Stack Overflow .
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            https://github.com/HiFi-LoFi/FFTConvolver.git

          • CLI

            gh repo clone HiFi-LoFi/FFTConvolver

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            git@github.com:HiFi-LoFi/FFTConvolver.git

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