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<!DOCTYPE html>
<html>
<head>
<meta charset="utf-8" />
<title>Test that decoder delay is handled</title>
<script src="/tests/SimpleTest/SimpleTest.js"></script>
<link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
</head>
<body>
<script class="testbody" type="text/javascript">
SimpleTest.waitForExplicitFinish();
const {AppConstants} =
SpecialPowers.ChromeUtils.import("resource://gre/modules/AppConstants.jsm");
var tests_half_a_second = [
"half-a-second-1ch-44100-aac.mp4",
"half-a-second-1ch-44100-flac.flac",
"half-a-second-1ch-44100-libmp3lame.mp3",
"half-a-second-1ch-44100-libopus.opus",
"half-a-second-1ch-44100-libopus.webm",
"half-a-second-1ch-44100-libvorbis.ogg",
"half-a-second-1ch-44100.wav",
"half-a-second-1ch-48000-aac.mp4",
"half-a-second-1ch-48000-flac.flac",
"half-a-second-1ch-48000-libmp3lame.mp3",
"half-a-second-1ch-48000-libopus.opus",
"half-a-second-1ch-48000-libopus.webm",
"half-a-second-1ch-48000-libvorbis.ogg",
"half-a-second-1ch-48000.wav",
"half-a-second-2ch-44100-aac.mp4",
"half-a-second-2ch-44100-flac.flac",
"half-a-second-2ch-44100-libmp3lame.mp3",
"half-a-second-2ch-44100-libopus.opus",
"half-a-second-2ch-44100-libopus.webm",
"half-a-second-2ch-44100-libvorbis.ogg",
"half-a-second-2ch-44100.wav",
"half-a-second-2ch-48000-aac.mp4",
"half-a-second-2ch-48000-flac.flac",
"half-a-second-2ch-48000-libmp3lame.mp3",
"half-a-second-2ch-48000-libopus.opus",
"half-a-second-2ch-48000-libopus.webm",
"half-a-second-2ch-48000-libvorbis.ogg",
"half-a-second-2ch-48000.wav",
];
// Those files are almost exactly half a second, but don't have enough pre-roll/padding
// information in the container, or the container isn't parsed properly, so
// aren't trimmed appropriately.
// vorbis webm, opus mp4, aac adts
var tests_adts = [
"half-a-second-1ch-44100-aac.aac",
"half-a-second-1ch-44100-libopus.mp4",
"half-a-second-1ch-44100-libvorbis.webm",
"half-a-second-1ch-48000-aac.aac",
"half-a-second-1ch-48000-libopus.mp4",
"half-a-second-1ch-48000-libvorbis.webm",
"half-a-second-2ch-44100-aac.aac",
"half-a-second-2ch-44100-libopus.mp4",
"half-a-second-2ch-44100-libvorbis.webm",
"half-a-second-2ch-48000-aac.aac",
"half-a-second-2ch-48000-libopus.mp4",
"half-a-second-2ch-48000-libvorbis.webm",
];
// Other files that have interesting characteristics.
var tests_others = [
{
// Very short VBR file, 16 frames of audio at 44100. Padding spanning two
// packets.
"path": "sixteen-frames.mp3",
"frameCount": 16,
"samplerate": 44100,
"fuzz": {}
},
{
// This is incorrect (the duration should be 0.5s exactly)
// This is tracked in https://github.com/mozilla/mp4parse-rust/issues/404
"path":"half-a-second-1ch-44100-aac-afconvert.mp4",
"frameCount": 22464,
"samplerate": 44100,
"fuzz": {
"android": 2
}
}
];
var all_tests = [tests_half_a_second, tests_adts, tests_others].flat();
var count = 0;
function checkDone() {
if (++count == all_tests.length) {
SimpleTest.finish();
}
}
async function doit() {
var context = new OfflineAudioContext(1, 128, 48000);
tests_half_a_second.forEach(async testfile => {
var response = await fetch(testfile);
var buffer = await response.arrayBuffer();
var decoded = await context.decodeAudioData(buffer);
is(
decoded.duration,
0.5,
"The file " + testfile + " is half a second."
);
// Value found empirically after looking at the files. The initial
// amplitude should be 0 at phase 0 because those files are sine wave.
// The compression is sometimes lossy and the first sample is not always
// exactly 0.0.
ok(
Math.abs(decoded.getChannelData(0)[0]) <= 0.022,
`The start point for ${testfile} is correct ${ decoded.getChannelData(0)[0] }`
);
checkDone();
});
tests_adts.forEach(async testfile => {
var response = await fetch(testfile);
var buffer = await response.arrayBuffer();
var decoded = await context.decodeAudioData(buffer);
// Value found empirically after looking at the files. ADTS containers
// don't have encoder delay / padding info so we can't trim correctly.
ok(
Math.abs(decoded.duration - 0.5) < 0.02,
`The ADTS file ${testfile} is about half a second (${decoded.duration}, error: ${Math.abs(decoded.duration-0.5)}).`
);
checkDone();
});
tests_others.forEach(async test => {
// Get an context at a specific rate to avoid duration changes due to resampling.
var contextAtRate = new OfflineAudioContext(1, 128, test.samplerate);
var response = await fetch(test.path);
var buffer = await response.arrayBuffer();
var decoded = await contextAtRate.decodeAudioData(buffer);
const fuzz = test.fuzz[AppConstants.platform] ?? 0;
ok(Math.abs(decoded.length - test.frameCount) <= fuzz, `${test.path} is ${decoded.length} frames long`);
checkDone();
});
}
doit();
</script>
</body>
</html>
|
