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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 17:32:43 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 17:32:43 +0000
commit6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch)
treea68f146d7fa01f0134297619fbe7e33db084e0aa /dom/media/webaudio/test/test_decoderDelay.html
parentInitial commit. (diff)
downloadthunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz
thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
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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>