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gst-plugins-rs/audio/audiofx/Cargo.toml

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Add audiofx plugin with audioecho element
2017-10-05 15:07:52 +00:00
[package]
name = "gst-plugin-audiofx"
Update versions to 0.6.0
2019-07-07 10:20:15 +00:00
version = "0.6.0"
Add audiofx plugin with audioecho element
2017-10-05 15:07:52 +00:00
authors = ["Sebastian Dröge <sebastian@centricular.com>"]
Change all links/mentions from gst-plugin-rs to gst-plugins-rs
2018-11-26 10:52:09 +00:00
repository = "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
Add audiofx plugin with audioecho element
2017-10-05 15:07:52 +00:00
license = "MIT/Apache-2.0"
Add build.rs that provides a version number, release date and other details
2019-06-03 10:53:58 +00:00
description = "Rust AudioFx Plugin"
audiofx: Add audioloudnorm filter based on the ffmpeg af_loudnorm filter This normalizes the loudness of an audio stream to a target loudness with a given maximum peak based on EBU R128. Conceptually it keeps a 3s lookahead for calculating the perceived loudness and based on that calculates the gain required to reach the target loudness. The calculated gains then go through a gaussian filter for smoothening and are then applied to the audio in 100ms blocks. Each of the 100ms blocks is then passed to a limiter filter to prevent going above the maximum peak. See http://k.ylo.ph/2016/04/04/loudnorm.html for some more details about the algorithm. It introduces 3s of latency and currently only works on 192kHz audio. Using it with a different sample rate requires resampling before and afterwards. The upsampling is required to calculate the true peak. Other than the ffmpeg filter it currently does not support two-pass processing but only one-pass/live processing. Compared to the ffmpeg filter this code was refactored considerably and the limiter implementation was fixed to actually work, as well as various other bugs in different places that were fixed.
2020-04-01 20:55:14 +00:00
edition = "2018"
Add audiofx plugin with audioecho element
2017-10-05 15:07:52 +00:00
[dependencies]
glib = { git = "https://github.com/gtk-rs/glib" }
Remove usage of subclassing feature, it's the default now
2019-09-09 07:45:06 +00:00
gstreamer = { git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs" }
gstreamer-base = { git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs" }
Change all references from GitHub to freedesktop.org GitLab
2018-11-03 18:24:21 +00:00
gstreamer-audio = { git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs" }
Update some dependencies
2019-05-24 11:33:51 +00:00
byte-slice-cast = "0.3"
Update to num-traits 0.2
2018-02-21 12:36:37 +00:00
num-traits = "0.2"
Update for new BaseTransformImpl::set_caps() return type
2019-11-20 22:06:16 +00:00
lazy_static = "1.0"
audiofx: Add audioloudnorm filter based on the ffmpeg af_loudnorm filter This normalizes the loudness of an audio stream to a target loudness with a given maximum peak based on EBU R128. Conceptually it keeps a 3s lookahead for calculating the perceived loudness and based on that calculates the gain required to reach the target loudness. The calculated gains then go through a gaussian filter for smoothening and are then applied to the audio in 100ms blocks. Each of the 100ms blocks is then passed to a limiter filter to prevent going above the maximum peak. See http://k.ylo.ph/2016/04/04/loudnorm.html for some more details about the algorithm. It introduces 3s of latency and currently only works on 192kHz audio. Using it with a different sample rate requires resampling before and afterwards. The upsampling is required to calculate the true peak. Other than the ffmpeg filter it currently does not support two-pass processing but only one-pass/live processing. Compared to the ffmpeg filter this code was refactored considerably and the limiter implementation was fixed to actually work, as well as various other bugs in different places that were fixed.
2020-04-01 20:55:14 +00:00
ebur128 = "0.1"
Add audiofx plugin with audioecho element
2017-10-05 15:07:52 +00:00
[lib]
name = "gstrsaudiofx"
audiofx: Add audioloudnorm filter based on the ffmpeg af_loudnorm filter This normalizes the loudness of an audio stream to a target loudness with a given maximum peak based on EBU R128. Conceptually it keeps a 3s lookahead for calculating the perceived loudness and based on that calculates the gain required to reach the target loudness. The calculated gains then go through a gaussian filter for smoothening and are then applied to the audio in 100ms blocks. Each of the 100ms blocks is then passed to a limiter filter to prevent going above the maximum peak. See http://k.ylo.ph/2016/04/04/loudnorm.html for some more details about the algorithm. It introduces 3s of latency and currently only works on 192kHz audio. Using it with a different sample rate requires resampling before and afterwards. The upsampling is required to calculate the true peak. Other than the ffmpeg filter it currently does not support two-pass processing but only one-pass/live processing. Compared to the ffmpeg filter this code was refactored considerably and the limiter implementation was fixed to actually work, as well as various other bugs in different places that were fixed.
2020-04-01 20:55:14 +00:00
crate-type = ["cdylib", "rlib"]
Add audiofx plugin with audioecho element
2017-10-05 15:07:52 +00:00
path = "src/lib.rs"
Add build.rs that provides a version number, release date and other details
2019-06-03 10:53:58 +00:00
audiofx: Add audioloudnorm filter based on the ffmpeg af_loudnorm filter This normalizes the loudness of an audio stream to a target loudness with a given maximum peak based on EBU R128. Conceptually it keeps a 3s lookahead for calculating the perceived loudness and based on that calculates the gain required to reach the target loudness. The calculated gains then go through a gaussian filter for smoothening and are then applied to the audio in 100ms blocks. Each of the 100ms blocks is then passed to a limiter filter to prevent going above the maximum peak. See http://k.ylo.ph/2016/04/04/loudnorm.html for some more details about the algorithm. It introduces 3s of latency and currently only works on 192kHz audio. Using it with a different sample rate requires resampling before and afterwards. The upsampling is required to calculate the true peak. Other than the ffmpeg filter it currently does not support two-pass processing but only one-pass/live processing. Compared to the ffmpeg filter this code was refactored considerably and the limiter implementation was fixed to actually work, as well as various other bugs in different places that were fixed.
2020-04-01 20:55:14 +00:00
[dev-dependencies]
gstreamer-check = { git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs" }
gstreamer-app = { git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs" }
Add build.rs that provides a version number, release date and other details
2019-06-03 10:53:58 +00:00
[build-dependencies]
Drop gst-plugin- prefix in plugin directory name
2020-04-05 14:38:52 +00:00
gst-plugin-version-helper = { path="../../version-helper" }
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