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| author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
|---|---|---|
| committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
| commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
| tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /dom/media/webaudio/test/test_sequentialBufferSourceWithResampling.html | |
| parent | Initial commit. (diff) | |
| download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip | |
Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
| -rw-r--r-- | dom/media/webaudio/test/test_sequentialBufferSourceWithResampling.html | 72 |
1 files changed, 72 insertions, 0 deletions
diff --git a/dom/media/webaudio/test/test_sequentialBufferSourceWithResampling.html b/dom/media/webaudio/test/test_sequentialBufferSourceWithResampling.html new file mode 100644 index 0000000000..5c03a8a911 --- /dev/null +++ b/dom/media/webaudio/test/test_sequentialBufferSourceWithResampling.html @@ -0,0 +1,72 @@ +<!DOCTYPE html> +<title>Test seamless playback of a series of resampled buffers</title> +<script src="/resources/testharness.js"></script> +<script src="/resources/testharnessreport.js"></script> +<script> +// Permitting some accumulation of rounding to int16_t. +// 64/2^15 would be only just small enough to detect off-by-one-subsample +// scheduling errors with the frequencies here. +const EPSILON = 4.0 / Math.pow(2, 15); +// Offsets test for rounding to nearest rather than up or down. +const OFFSETS = [EPSILON, 1.0 - EPSILON]; +// The ratio of resampling is 147:160, so 256 start points is enough to cover +// every fractional offset. +const LENGTH = 256; + +function do_test(context_rate, buffer_rate, start_offset) { + + var context = + new OfflineAudioContext(2, LENGTH, context_rate); + + var merger = context.createChannelMerger(context.destination.channelCount); + merger.connect(context.destination); + + // Create an audio signal that will be repeated + var repeating_signal = context.createBuffer(1, 1, buffer_rate); + repeating_signal.getChannelData(0)[0] = 0.5; + + // Schedule a series of nodes to repeat the signal. + for (var i = 0; i < LENGTH; ++i) { + var source = context.createBufferSource(); + source.buffer = repeating_signal; + source.connect(merger, 0, 0); + source.start((i + start_offset) / buffer_rate); + } + + // A single long signal should produce the same result. + var long_signal = context.createBuffer(1, LENGTH, buffer_rate); + var c = long_signal.getChannelData(0); + for (var i = 0; i < c.length; ++i) { + c[i] = 0.5; + } + + var source = context.createBufferSource(); + source.buffer = long_signal; + source.connect(merger, 0, 1); + source.start(start_offset / buffer_rate); + + return context.startRendering(). + then((buffer) => { + series_output = buffer.getChannelData(0); + expected = buffer.getChannelData(1); + + for (var i = 0; i < buffer.length; ++i) { + assert_approx_equals(series_output[i], expected[i], EPSILON, + "series output at " + i); + } + }); +} + +function start_tests(context_rate, buffer_rate) { + OFFSETS.forEach((start_offset) => { + promise_test(() => do_test(context_rate, buffer_rate, start_offset), + "" + context_rate + " context, " + + buffer_rate + " buffer, " + + start_offset + " start"); + }); +} + +start_tests(48000, 44100); +start_tests(44100, 48000); + +</script> |