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Commit f18e6ccd authored by Raymond Toy's avatar Raymond Toy Committed by Commit Bot
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Remove Dezippering from the DelayNode 

Remove dezippering from delayTime.  The value will now change
immediately instead of gradually changing from the old to new
value.

Chromium Feature: https://www.chromestatus.com/features/5287995770929152
Intent to ship: https://groups.google.com/a/chromium.org/d/msg/blink-dev/YKYRrh0nWMo/aGzd3049AgAJ

Bug: 550533
Test: Delay/dezipper.html
Change-Id: Id3ce76401d4342205c5d67079f6684514ae7e2a2
Reviewed-on: https://chromium-review.googlesource.com/609443
Commit-Queue: Raymond Toy <rtoy@chromium.org>
Reviewed-by: default avatarHongchan Choi <hongchan@chromium.org>
Cr-Commit-Position: refs/heads/master@{#532682}
parent d86c522b
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Showing with 185 additions and 28 deletions
third_party/WebKit/LayoutTests/webaudio/Delay/no-dezippering.html 0 → 100644
View file @ f18e6ccd
<!DOCTYPE html>
<html>
<head>
<title>
Test DelayNode Has No Dezippering
</title>
<script src="../../resources/testharness.js"></script>
<script src="../../resources/testharnessreport.js"></script>
<script src="../resources/audit-util.js"></script>
<script src="../resources/audit.js"></script>
</head>
<body>
<script id="layout-test-code">
// The sample rate must be a power of two to avoid any round-off errors in
// computing when to suspend a context on a rendering quantum boundary.
// Otherwise this is pretty arbitrary.
let sampleRate = 16384;
let audit = Audit.createTaskRunner();
audit.define(
{label: 'test0', description: 'Test DelayNode has no dezippering'},
(task, should) => {
let context = new OfflineAudioContext(1, sampleRate, sampleRate);
// Simple integer ramp for testing delay node
let buffer = new AudioBuffer(
{length: context.length, sampleRate: context.sampleRate});
let rampData = buffer.getChannelData(0);
for (let k = 0; k < rampData.length; ++k) {
rampData[k] = k + 1;
}
// |delay0Frame| is the initial delay in frames. |delay1Frame| is
// the new delay in frames. These must be integers.
let delay0Frame = 64;
let delay1Frame = 16;
let src = new AudioBufferSourceNode(context, {buffer: buffer});
let delay = new DelayNode(
context, {delayTime: delay0Frame / context.sampleRate});
src.connect(delay).connect(context.destination);
// After a render quantum, change the delay to |delay1Frame|.
context.suspend(RENDER_QUANTUM_FRAMES / context.sampleRate)
.then(() => {
delay.delayTime.value = delay1Frame / context.sampleRate;
})
.then(() => context.resume());
src.start();
context.startRendering()
.then(renderedBuffer => {
let renderedData = renderedBuffer.getChannelData(0);
// The first |delay0Frame| frames should be zero.
should(
renderedData.slice(0, delay0Frame),
'output[0:' + (delay0Frame - 1) + ']')
.beConstantValueOf(0);
// Now we have the ramp should show up from the delay.
let ramp0 =
new Float32Array(RENDER_QUANTUM_FRAMES - delay0Frame);
for (let k = 0; k < ramp0.length; ++k) {
ramp0[k] = rampData[k];
}
should(
renderedData.slice(delay0Frame, RENDER_QUANTUM_FRAMES),
'output[' + delay0Frame + ':' +
(RENDER_QUANTUM_FRAMES - 1) + ']')
.beEqualToArray(ramp0);
// After one rendering quantum, the delay is changed to
// |delay1Frame|.
let ramp1 =
new Float32Array(context.length - RENDER_QUANTUM_FRAMES);
for (let k = 0; k < ramp1.length; ++k) {
// ramp1[k] = 1 + k + RENDER_QUANTUM_FRAMES - delay1Frame;
ramp1[k] =
rampData[k + RENDER_QUANTUM_FRAMES - delay1Frame];
}
should(
renderedData.slice(RENDER_QUANTUM_FRAMES),
'output[' + RENDER_QUANTUM_FRAMES + ':]')
.beEqualToArray(ramp1);
})
.then(() => task.done());
});
audit.define(
{label: 'test1', description: 'Test value setter and setValueAtTime'},
(task, should) => {
testWithAutomation(should, {prefix: ''}).then(() => task.done());
});
audit.define(
{label: 'test2', description: 'Test value setter and modulation'},
(task, should) => {
testWithAutomation(should, {
prefix: 'With modulation: ',
modulator: true
}).then(() => task.done());
});
// Compare .value setter with setValueAtTime, Optionally allow modulation
// of |delayTime|.
function testWithAutomation(should, options) {
let prefix = options.prefix;
// Channel 0 is the output of delay node using the setter and channel 1
// is the output using setValueAtTime.
let context = new OfflineAudioContext(2, sampleRate, sampleRate);
let merger = new ChannelMergerNode(
context, {numberOfInputs: context.destination.channelCount});
merger.connect(context.destination);
let src = new OscillatorNode(context);
// |delay0Frame| is the initial delay value in frames. |delay1Frame| is
// the new delay in frames. The values here are constrained only by the
// constraints for a DelayNode. These are pretty arbitrary except we
// wanted them to be fractional so as not be on a frame boundary to
// test interpolation compared with |setValueAtTime()|..
let delay0Frame = 3.1;
let delay1Frame = 47.2;
let delayTest = new DelayNode(
context, {delayTime: delay0Frame / context.sampleRate});
let delayRef = new DelayNode(
context, {delayTime: delay0Frame / context.sampleRate});
src.connect(delayTest).connect(merger, 0, 0);
src.connect(delayRef).connect(merger, 0, 1);
if (options.modulator) {
// Fairly arbitrary modulation of the delay time, with a peak
// variation of 10 ms.
let mod = new OscillatorNode(context, {frequency: 1000});
let modGain = new GainNode(context, {gain: .01});
mod.connect(modGain);
modGain.connect(delayTest.delayTime);
modGain.connect(delayRef.delayTime);
mod.start();
}
// The time at which the delay time of |delayTest| node will be
// changed. This MUST be on a render quantum boundary, but is
// otherwise arbitrary.
let changeTime = 3 * RENDER_QUANTUM_FRAMES / context.sampleRate;
// Schedule the delay change on |delayRef| and also apply the value
// setter for |delayTest| at |changeTime|.
delayRef.delayTime.setValueAtTime(
delay1Frame / context.sampleRate, changeTime);
context.suspend(changeTime)
.then(() => {
delayTest.delayTime.value = delay1Frame / context.sampleRate;
})
.then(() => context.resume());
src.start();
return context.startRendering().then(renderedBuffer => {
let actual = renderedBuffer.getChannelData(0);
let expected = renderedBuffer.getChannelData(1);
let match = should(actual, prefix + '.value setter output')
.beEqualToArray(expected);
should(
match,
prefix + '.value setter output matches setValueAtTime output')
.beTrue();
});
}
audit.run();
</script>
</body>
</html>
third_party/WebKit/Source/modules/webaudio/DelayDSPKernel.cpp
View file @ f18e6ccd
...@@ -30,8 +30,6 @@ ...@@ -30,8 +30,6 @@
namespace blink { namespace blink {
const float kSmoothingTimeConstant = 0.020f; // 20ms
DelayDSPKernel::DelayDSPKernel(DelayProcessor* processor) DelayDSPKernel::DelayDSPKernel(DelayProcessor* processor)
: AudioDelayDSPKernel(processor, AudioUtilities::kRenderQuantumFrames) { : AudioDelayDSPKernel(processor, AudioUtilities::kRenderQuantumFrames) {
DCHECK(processor); DCHECK(processor);
...@@ -48,9 +46,6 @@ DelayDSPKernel::DelayDSPKernel(DelayProcessor* processor) ...@@ -48,9 +46,6 @@ DelayDSPKernel::DelayDSPKernel(DelayProcessor* processor)
buffer_.Allocate( buffer_.Allocate(
BufferLengthForDelay(max_delay_time_, processor->SampleRate())); BufferLengthForDelay(max_delay_time_, processor->SampleRate()));
buffer_.Zero(); buffer_.Zero();
smoothing_rate_ = AudioUtilities::DiscreteTimeConstantForSampleRate(
kSmoothingTimeConstant, processor->SampleRate());
} }
bool DelayDSPKernel::HasSampleAccurateValues() { bool DelayDSPKernel::HasSampleAccurateValues() {
......
third_party/WebKit/Source/platform/audio/AudioDelayDSPKernel.cpp
View file @ f18e6ccd
...@@ -31,21 +31,17 @@ ...@@ -31,21 +31,17 @@
namespace blink { namespace blink {
const float kSmoothingTimeConstant = 0.020f; // 20ms
AudioDelayDSPKernel::AudioDelayDSPKernel(AudioDSPKernelProcessor* processor, AudioDelayDSPKernel::AudioDelayDSPKernel(AudioDSPKernelProcessor* processor,
size_t processing_size_in_frames) size_t processing_size_in_frames)
: AudioDSPKernel(processor), : AudioDSPKernel(processor),
write_index_(0), write_index_(0),
first_time_(true),
delay_times_(processing_size_in_frames) {} delay_times_(processing_size_in_frames) {}
AudioDelayDSPKernel::AudioDelayDSPKernel(double max_delay_time, AudioDelayDSPKernel::AudioDelayDSPKernel(double max_delay_time,
float sample_rate) float sample_rate)
: AudioDSPKernel(sample_rate), : AudioDSPKernel(sample_rate),
max_delay_time_(max_delay_time), max_delay_time_(max_delay_time),
write_index_(0), write_index_(0) {
first_time_(true) {
DCHECK_GT(max_delay_time, 0.0); DCHECK_GT(max_delay_time, 0.0);
DCHECK(!std::isnan(max_delay_time)); DCHECK(!std::isnan(max_delay_time));
if (max_delay_time <= 0.0 || std::isnan(max_delay_time)) if (max_delay_time <= 0.0 || std::isnan(max_delay_time))
...@@ -58,9 +54,6 @@ AudioDelayDSPKernel::AudioDelayDSPKernel(double max_delay_time, ...@@ -58,9 +54,6 @@ AudioDelayDSPKernel::AudioDelayDSPKernel(double max_delay_time,
buffer_.Allocate(buffer_length); buffer_.Allocate(buffer_length);
buffer_.Zero(); buffer_.Zero();
smoothing_rate_ = AudioUtilities::DiscreteTimeConstantForSampleRate(
kSmoothingTimeConstant, sample_rate);
} }
size_t AudioDelayDSPKernel::BufferLengthForDelay(double max_delay_time, size_t AudioDelayDSPKernel::BufferLengthForDelay(double max_delay_time,
...@@ -112,11 +105,6 @@ void AudioDelayDSPKernel::Process(const float* source, ...@@ -112,11 +105,6 @@ void AudioDelayDSPKernel::Process(const float* source,
// Make sure the delay time is in a valid range. // Make sure the delay time is in a valid range.
delay_time = clampTo(delay_time, 0.0, max_time); delay_time = clampTo(delay_time, 0.0, max_time);
if (first_time_) {
current_delay_time_ = delay_time;
first_time_ = false;
}
} }
for (unsigned i = 0; i < frames_to_process; ++i) { for (unsigned i = 0; i < frames_to_process; ++i) {
...@@ -126,14 +114,9 @@ void AudioDelayDSPKernel::Process(const float* source, ...@@ -126,14 +114,9 @@ void AudioDelayDSPKernel::Process(const float* source,
delay_time = max_time; delay_time = max_time;
else else
delay_time = clampTo(delay_time, 0.0, max_time); delay_time = clampTo(delay_time, 0.0, max_time);
current_delay_time_ = delay_time;
} else {
// Approach desired delay time.
current_delay_time_ +=
(delay_time - current_delay_time_) * smoothing_rate_;
} }
double desired_delay_frames = current_delay_time_ * sample_rate; double desired_delay_frames = delay_time * sample_rate;
double read_position = write_index_ + buffer_length - desired_delay_frames; double read_position = write_index_ + buffer_length - desired_delay_frames;
if (read_position >= buffer_length) if (read_position >= buffer_length)
...@@ -159,7 +142,6 @@ void AudioDelayDSPKernel::Process(const float* source, ...@@ -159,7 +142,6 @@ void AudioDelayDSPKernel::Process(const float* source,
} }
void AudioDelayDSPKernel::Reset() { void AudioDelayDSPKernel::Reset() {
first_time_ = true;
buffer_.Zero(); buffer_.Zero();
} }
......
third_party/WebKit/Source/platform/audio/AudioDelayDSPKernel.h
View file @ f18e6ccd
...@@ -61,9 +61,6 @@ class PLATFORM_EXPORT AudioDelayDSPKernel : public AudioDSPKernel { ...@@ -61,9 +61,6 @@ class PLATFORM_EXPORT AudioDelayDSPKernel : public AudioDSPKernel {
AudioFloatArray buffer_; AudioFloatArray buffer_;
double max_delay_time_; double max_delay_time_;
int write_index_; int write_index_;
double current_delay_time_;
double smoothing_rate_;
bool first_time_;
double desired_delay_frames_; double desired_delay_frames_;
AudioFloatArray delay_times_; AudioFloatArray delay_times_;
......
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