1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
|
/*
* Copyright (C) 2010 Google Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
* WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
* DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
* DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
* ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
* THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "config.h"
#if ENABLE(WEB_AUDIO)
#include "AudioParam.h"
#include "AudioNode.h"
#include "AudioNodeOutput.h"
#include "AudioUtilities.h"
#include "FloatConversion.h"
#include <wtf/MathExtras.h>
namespace WebCore {
const double AudioParam::DefaultSmoothingConstant = 0.05;
const double AudioParam::SnapThreshold = 0.001;
float AudioParam::value()
{
// Update value for timeline.
if (context().isAudioThread()) {
bool hasValue;
float timelineValue = m_timeline.valueForContextTime(context(), narrowPrecisionToFloat(m_value), hasValue);
if (hasValue)
m_value = timelineValue;
}
return narrowPrecisionToFloat(m_value);
}
void AudioParam::setValue(float value)
{
// Check against JavaScript giving us bogus floating-point values.
// Don't ASSERT, since this can happen if somebody writes bad JS.
if (!std::isnan(value) && !std::isinf(value))
m_value = value;
}
float AudioParam::smoothedValue()
{
return narrowPrecisionToFloat(m_smoothedValue);
}
bool AudioParam::smooth()
{
// If values have been explicitly scheduled on the timeline, then use the exact value.
// Smoothing effectively is performed by the timeline.
bool useTimelineValue = false;
m_value = m_timeline.valueForContextTime(context(), narrowPrecisionToFloat(m_value), useTimelineValue);
if (m_smoothedValue == m_value) {
// Smoothed value has already approached and snapped to value.
return true;
}
if (useTimelineValue)
m_smoothedValue = m_value;
else {
// Dezipper - exponential approach.
m_smoothedValue += (m_value - m_smoothedValue) * m_smoothingConstant;
// If we get close enough then snap to actual value.
if (fabs(m_smoothedValue - m_value) < SnapThreshold) // FIXME: the threshold needs to be adjustable depending on range - but this is OK general purpose value.
m_smoothedValue = m_value;
}
return false;
}
float AudioParam::finalValue()
{
float value;
calculateFinalValues(&value, 1, false);
return value;
}
void AudioParam::calculateSampleAccurateValues(float* values, unsigned numberOfValues)
{
bool isSafe = context().isAudioThread() && values && numberOfValues;
ASSERT(isSafe);
if (!isSafe)
return;
calculateFinalValues(values, numberOfValues, true);
}
void AudioParam::calculateFinalValues(float* values, unsigned numberOfValues, bool sampleAccurate)
{
bool isGood = context().isAudioThread() && values && numberOfValues;
ASSERT(isGood);
if (!isGood)
return;
// The calculated result will be the "intrinsic" value summed with all audio-rate connections.
if (sampleAccurate) {
// Calculate sample-accurate (a-rate) intrinsic values.
calculateTimelineValues(values, numberOfValues);
} else {
// Calculate control-rate (k-rate) intrinsic value.
bool hasValue;
float timelineValue = m_timeline.valueForContextTime(context(), narrowPrecisionToFloat(m_value), hasValue);
if (hasValue)
m_value = timelineValue;
values[0] = narrowPrecisionToFloat(m_value);
}
// Now sum all of the audio-rate connections together (unity-gain summing junction).
// Note that connections would normally be mono, but we mix down to mono if necessary.
RefPtr<AudioBus> summingBus = AudioBus::create(1, numberOfValues, false);
summingBus->setChannelMemory(0, values, numberOfValues);
for (auto& output : m_renderingOutputs) {
ASSERT(output);
// Render audio from this output.
AudioBus* connectionBus = output->pull(0, AudioNode::ProcessingSizeInFrames);
// Sum, with unity-gain.
summingBus->sumFrom(*connectionBus);
}
}
void AudioParam::calculateTimelineValues(float* values, unsigned numberOfValues)
{
// Calculate values for this render quantum.
// Normally numberOfValues will equal AudioNode::ProcessingSizeInFrames (the render quantum size).
double sampleRate = context().sampleRate();
double startTime = context().currentTime();
double endTime = startTime + numberOfValues / sampleRate;
// Note we're running control rate at the sample-rate.
// Pass in the current value as default value.
m_value = m_timeline.valuesForTimeRange(startTime, endTime, narrowPrecisionToFloat(m_value), values, numberOfValues, sampleRate, sampleRate);
}
void AudioParam::connect(AudioNodeOutput* output)
{
ASSERT(context().isGraphOwner());
ASSERT(output);
if (!output)
return;
if (!m_outputs.add(output).isNewEntry)
return;
output->addParam(this);
changedOutputs();
}
void AudioParam::disconnect(AudioNodeOutput* output)
{
ASSERT(context().isGraphOwner());
ASSERT(output);
if (!output)
return;
if (m_outputs.remove(output)) {
changedOutputs();
output->removeParam(this);
}
}
} // namespace WebCore
#endif // ENABLE(WEB_AUDIO)
|
