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| author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
|---|---|---|
| committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
| commit | 43a97878ce14b72f0981164f87f2e35e14151312 (patch) | |
| tree | 620249daf56c0258faa40cbdcf9cfba06de2a846 /testing/web-platform/tests/webaudio/the-audio-api/the-audiobuffersourcenode-interface/audiobuffersource-playbackrate-zero.html | |
| parent | Initial commit. (diff) | |
| download | firefox-43a97878ce14b72f0981164f87f2e35e14151312.tar.xz firefox-43a97878ce14b72f0981164f87f2e35e14151312.zip | |
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'testing/web-platform/tests/webaudio/the-audio-api/the-audiobuffersourcenode-interface/audiobuffersource-playbackrate-zero.html')
| -rw-r--r-- | testing/web-platform/tests/webaudio/the-audio-api/the-audiobuffersourcenode-interface/audiobuffersource-playbackrate-zero.html | 116 |
1 files changed, 116 insertions, 0 deletions
diff --git a/testing/web-platform/tests/webaudio/the-audio-api/the-audiobuffersourcenode-interface/audiobuffersource-playbackrate-zero.html b/testing/web-platform/tests/webaudio/the-audio-api/the-audiobuffersourcenode-interface/audiobuffersource-playbackrate-zero.html new file mode 100644 index 0000000000..5624054e32 --- /dev/null +++ b/testing/web-platform/tests/webaudio/the-audio-api/the-audiobuffersourcenode-interface/audiobuffersource-playbackrate-zero.html @@ -0,0 +1,116 @@ +<!DOCTYPE html> +<html> + <head> + <title> + audiobuffersource-playbackrate-zero.html + </title> + <script src="/resources/testharness.js"></script> + <script src="/resources/testharnessreport.js"></script> + <script src="/webaudio/resources/audit-util.js"></script> + <script src="/webaudio/resources/audit.js"></script> + </head> + <body> + <script id="layout-test-code"> + // Sample rate should be power of 128 to observe the change of AudioParam + // at the beginning of rendering quantum. (playbackRate is k-rate) This is + // the minimum sample rate in the valid sample rate range. + let sampleRate = 8192; + + // The render duration in seconds, and the length in samples. + let renderDuration = 1.0; + let renderLength = renderDuration * sampleRate; + + let context = new OfflineAudioContext(1, renderLength, sampleRate); + let audit = Audit.createTaskRunner(); + + + // Task: Render the actual buffer and compare with the reference. + audit.define('synthesize-verify', (task, should) => { + let ramp = context.createBufferSource(); + let rampBuffer = createLinearRampBuffer(context, renderLength); + ramp.buffer = rampBuffer; + + ramp.connect(context.destination); + ramp.start(); + + // Leave the playbackRate as 1 for the first half, then change it + // to zero at the exact half. The zero playback rate should hold the + // sample value of the buffer index at the moment. (sample-and-hold) + ramp.playbackRate.setValueAtTime(1.0, 0.0); + ramp.playbackRate.setValueAtTime(0.0, renderDuration / 2); + + context.startRendering() + .then(function(renderedBuffer) { + let data = renderedBuffer.getChannelData(0); + let rampData = rampBuffer.getChannelData(0); + let half = rampData.length / 2; + let passed = true; + let i; + + for (i = 1; i < rampData.length; i++) { + if (i < half) { + // Before the half position, the actual should match with the + // original ramp data. + if (data[i] !== rampData[i]) { + passed = false; + break; + } + } else { + // From the half position, the actual value should not change. + if (data[i] !== rampData[half]) { + passed = false; + break; + } + } + } + + should(passed, 'The zero playbackRate') + .message( + 'held the sample value correctly', + 'should hold the sample value. ' + + 'Expected ' + rampData[half] + ' but got ' + data[i] + + ' at the index ' + i); + }) + .then(() => task.done()); + }); + + audit.define('subsample start with playback rate 0', (task, should) => { + let context = new OfflineAudioContext(1, renderLength, sampleRate); + let rampBuffer = new AudioBuffer( + {length: renderLength, sampleRate: context.sampleRate}); + let data = new Float32Array(renderLength); + let startValue = 5; + for (let k = 0; k < data.length; ++k) { + data[k] = k + startValue; + } + rampBuffer.copyToChannel(data, 0); + + let src = new AudioBufferSourceNode( + context, {buffer: rampBuffer, playbackRate: 0}); + + src.connect(context.destination); + + // Purposely start the source between frame boundaries + let startFrame = 27.3; + src.start(startFrame / context.sampleRate); + + context.startRendering() + .then(audioBuffer => { + let actualStartFrame = Math.ceil(startFrame); + let audio = audioBuffer.getChannelData(0); + + should( + audio.slice(0, actualStartFrame), + `output[0:${actualStartFrame - 1}]`) + .beConstantValueOf(0); + should( + audio.slice(actualStartFrame), `output[${actualStartFrame}:]`) + .beConstantValueOf(startValue); + }) + .then(() => task.done()); + }); + + audit.run(); + </script> + </body> +</html> |