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| author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
|---|---|---|
| committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
| commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
| tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /dom/media/webaudio/test/test_decoderDelay.html | |
| parent | Initial commit. (diff) | |
| download | thunderbird-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>
Diffstat (limited to 'dom/media/webaudio/test/test_decoderDelay.html')
| -rw-r--r-- | dom/media/webaudio/test/test_decoderDelay.html | 144 |
1 files changed, 144 insertions, 0 deletions
diff --git a/dom/media/webaudio/test/test_decoderDelay.html b/dom/media/webaudio/test/test_decoderDelay.html new file mode 100644 index 0000000000..d0fbfbed29 --- /dev/null +++ b/dom/media/webaudio/test/test_decoderDelay.html @@ -0,0 +1,144 @@ +<!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> |