gst-plugins-rs/audio/audiofx/tests/audioloudnorm.rs

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// Copyright (C) 2020 Sebastian Dröge <sebastian@centricular.com>
//
// This Source Code Form is subject to the terms of the Mozilla Public License, v2.0.
// If a copy of the MPL was not distributed with this file, You can obtain one at
// <https://mozilla.org/MPL/2.0/>.
//
// SPDX-License-Identifier: MPL-2.0
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use gst::prelude::*;
use byte_slice_cast::*;
use std::sync::{Arc, Mutex};
fn init() {
use std::sync::Once;
static INIT: Once = Once::new();
INIT.call_once(|| {
gst::init().unwrap();
gstrsaudiofx::plugin_register_static().expect("Failed to register rsaudiofx plugin");
});
}
fn run_test(
first_input: &str,
second_input: Option<&str>,
num_buffers: u32,
samples_per_buffer: u32,
channels: u32,
expected_loudness: f64,
) {
init();
let format = if cfg!(target_endian = "little") {
format!("audio/x-raw,format=F64LE,rate=192000,channels={}", channels)
} else {
format!("audio/x-raw,format=F64BE,rate=192000,channels={}", channels)
};
let pipeline = if let Some(second_input) = second_input {
gst::parse_launch(&format!(
"audiotestsrc {first_input} num-buffers={num_buffers} samplesperbuffer={samples_per_buffer} ! {format} ! audiomixer name=mixer output-buffer-duration={output_buffer_duration} ! {format} ! rsaudioloudnorm ! appsink name=sink audiotestsrc {second_input} num-buffers={num_buffers} samplesperbuffer={samples_per_buffer} ! {format} ! mixer.",
first_input = first_input,
second_input = second_input,
num_buffers = num_buffers,
samples_per_buffer = samples_per_buffer,
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output_buffer_duration = samples_per_buffer as u64 * *gst::ClockTime::SECOND / 192_000,
format = format,
))
} else {
gst::parse_launch(&format!(
"audiotestsrc {first_input} num-buffers={num_buffers} samplesperbuffer={samples_per_buffer} ! {format} ! rsaudioloudnorm ! appsink name=sink",
first_input = first_input,
num_buffers = num_buffers,
samples_per_buffer = samples_per_buffer,
format = format,
))
}
.unwrap()
.downcast::<gst::Pipeline>()
.unwrap();
let sink = pipeline
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.by_name("sink")
.unwrap()
.downcast::<gst_app::AppSink>()
.unwrap();
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sink.set_sync(false);
let caps = gst_audio::AudioInfo::builder(gst_audio::AUDIO_FORMAT_F64, 192_000, channels)
.build()
.unwrap()
.to_caps()
.unwrap();
sink.set_caps(Some(&caps));
let samples = Arc::new(Mutex::new(Vec::new()));
let samples_clone = samples.clone();
sink.set_callbacks(
gst_app::AppSinkCallbacks::builder()
.new_sample(move |sink| {
let sample = sink.pull_sample().unwrap();
let mut samples = samples_clone.lock().unwrap();
samples.push(sample);
Ok(gst::FlowSuccess::Ok)
})
.build(),
);
pipeline.set_state(gst::State::Playing).unwrap();
let mut eos = false;
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let bus = pipeline.bus().unwrap();
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while let Some(msg) = bus.timed_pop(gst::ClockTime::NONE) {
use gst::MessageView;
match msg.view() {
MessageView::Eos(..) => {
eos = true;
break;
}
MessageView::Error(..) => unreachable!(),
_ => (),
}
}
pipeline.set_state(gst::State::Null).unwrap();
assert!(eos);
let samples = samples.lock().unwrap();
let mut r128 = ebur128::EbuR128::new(
channels,
192_000,
ebur128::Mode::I | ebur128::Mode::SAMPLE_PEAK,
)
.unwrap();
let mut num_samples = 0;
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let mut expected_ts = gst::ClockTime::ZERO;
for sample in samples.iter() {
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use std::cmp::Ordering;
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let buf = sample.buffer().unwrap();
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let ts = buf.pts().expect("undefined pts");
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match ts.cmp(&expected_ts) {
Ordering::Greater => {
assert!(
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ts - expected_ts <= gst::ClockTime::NSECOND,
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"TS is {} instead of {}",
ts,
expected_ts
);
}
Ordering::Less => {
assert!(
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expected_ts - ts <= gst::ClockTime::NSECOND,
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"TS is {} instead of {}",
ts,
expected_ts
);
}
Ordering::Equal => (),
}
let map = buf.map_readable().unwrap();
let data = map.as_slice_of::<f64>().unwrap();
num_samples += data.len() / channels as usize;
r128.add_frames_f64(data).unwrap();
expected_ts += (data.len() as u64 / channels as u64).seconds() / 192_000;
}
assert_eq!(
num_samples,
num_buffers as usize * samples_per_buffer as usize
);
let loudness = r128.loudness_global().unwrap();
if expected_loudness.classify() == std::num::FpCategory::Infinite && expected_loudness < 0.0 {
assert!(
loudness.classify() == std::num::FpCategory::Infinite && loudness < 0.0,
"Loudness is {} instead of {}",
loudness,
expected_loudness,
);
} else {
assert!(
f64::abs(loudness - expected_loudness) < 1.0,
"Loudness is {} instead of {}",
loudness,
expected_loudness,
);
}
for c in 0..channels {
let peak = 20.0 * f64::log10(r128.sample_peak(c).unwrap());
assert!(
peak <= -2.0,
"Peak {} for channel {} is above -2.0",
c,
peak,
);
}
}
#[test]
fn basic() {
run_test("wave=sine", None, 1000, 1920, 1, -24.0);
}
#[test]
fn basic_white_noise() {
run_test("wave=white-noise", None, 1000, 1920, 1, -24.0);
}
#[test]
fn remaining_at_eos() {
run_test("wave=sine", None, 1000, 1024, 1, -24.0);
}
#[test]
fn short_input() {
run_test("wave=sine", None, 100, 1024, 1, -24.0);
}
#[test]
fn basic_two_channels() {
run_test("wave=sine", None, 1000, 1920, 2, -24.0);
}
#[test]
fn silence() {
run_test("wave=silence", None, 1000, 1024, 1, std::f64::NEG_INFINITY);
}
#[test]
fn quiet() {
// -6dB
run_test("wave=sine volume=0.5", None, 1000, 1024, 1, -24.0);
}
#[test]
fn very_quiet() {
// -20dB
run_test("wave=sine volume=0.1", None, 1000, 1024, 1, -24.0);
}
#[test]
fn very_very_quiet() {
// -40dB
run_test("wave=sine volume=0.01", None, 1000, 1024, 1, -24.0);
}
#[test]
fn below_threshold() {
// -70dB
run_test(
"wave=sine volume=0.00045",
None,
1000,
1024,
1,
std::f64::NEG_INFINITY,
);
}
#[test]
fn limiter() {
run_test(
"wave=sine volume=0.05",
Some("wave=ticks sine-periods-per-tick=1 tick-interval=4000000000"),
1000,
1024,
1,
-24.0,
);
}
#[test]
fn limiter_on_first_frame() {
run_test(
"wave=sine volume=0.05",
Some("wave=ticks sine-periods-per-tick=10 tick-interval=4000000000"),
1000,
1024,
1,
-24.0,
);
}