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
|
/*
* Copyright (C) 2012, 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 INC. 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 INC. 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) && ENABLE(MEDIA_STREAM)
#include "MediaStreamAudioSourceNode.h"
#include "AudioContext.h"
#include "AudioNodeOutput.h"
#include "AudioSourceProvider.h"
#include "AudioUtilities.h"
#include "Logging.h"
#include "MediaStreamAudioSourceOptions.h"
#include "WebAudioSourceProvider.h"
#include <wtf/IsoMallocInlines.h>
#include <wtf/Locker.h>
namespace WebCore {
WTF_MAKE_ISO_ALLOCATED_IMPL(MediaStreamAudioSourceNode);
ExceptionOr<Ref<MediaStreamAudioSourceNode>> MediaStreamAudioSourceNode::create(BaseAudioContext& context, MediaStreamAudioSourceOptions&& options)
{
RELEASE_ASSERT(options.mediaStream);
auto audioTracks = options.mediaStream->getAudioTracks();
if (audioTracks.isEmpty())
return Exception { InvalidStateError, "Media stream has no audio tracks"_s };
RefPtr<WebAudioSourceProvider> provider;
for (auto& track : audioTracks) {
provider = track->createAudioSourceProvider();
if (provider)
break;
}
if (!provider)
return Exception { InvalidStateError, "Could not find an audio track with an audio source provider"_s };
auto node = adoptRef(*new MediaStreamAudioSourceNode(context, *options.mediaStream, provider.releaseNonNull()));
node->setFormat(2, context.sampleRate());
// Context keeps reference until node is disconnected.
context.sourceNodeWillBeginPlayback(node);
return node;
}
MediaStreamAudioSourceNode::MediaStreamAudioSourceNode(BaseAudioContext& context, MediaStream& mediaStream, Ref<WebAudioSourceProvider>&& provider)
: AudioNode(context, NodeTypeMediaStreamAudioSource)
, m_mediaStream(mediaStream)
, m_provider(provider)
{
m_provider->setClient(this);
// Default to stereo. This could change depending on the format of the MediaStream's audio track.
addOutput(2);
initialize();
}
MediaStreamAudioSourceNode::~MediaStreamAudioSourceNode()
{
m_provider->setClient(nullptr);
uninitialize();
}
void MediaStreamAudioSourceNode::setFormat(size_t numberOfChannels, float sourceSampleRate)
{
// Synchronize with process().
Locker locker { m_processLock };
if (numberOfChannels == m_sourceNumberOfChannels && sourceSampleRate == m_sourceSampleRate)
return;
// The sample-rate must be equal to the context's sample-rate.
if (!numberOfChannels || numberOfChannels > AudioContext::maxNumberOfChannels) {
// process() will generate silence for these uninitialized values.
LOG(Media, "MediaStreamAudioSourceNode::setFormat(%u, %f) - unhandled format change", static_cast<unsigned>(numberOfChannels), sourceSampleRate);
m_sourceNumberOfChannels = 0;
return;
}
m_sourceNumberOfChannels = numberOfChannels;
m_sourceSampleRate = sourceSampleRate;
float sampleRate = this->sampleRate();
if (sourceSampleRate == sampleRate)
m_multiChannelResampler = nullptr;
else {
double scaleFactor = sourceSampleRate / sampleRate;
m_multiChannelResampler = makeUnique<MultiChannelResampler>(scaleFactor, numberOfChannels, AudioUtilities::renderQuantumSize, std::bind(&MediaStreamAudioSourceNode::provideInput, this, std::placeholders::_1, std::placeholders::_2));
}
m_sourceNumberOfChannels = numberOfChannels;
{
// The context must be locked when changing the number of output channels.
Locker contextLocker { context().graphLock() };
// Do any necesssary re-configuration to the output's number of channels.
output(0)->setNumberOfChannels(numberOfChannels);
}
}
void MediaStreamAudioSourceNode::provideInput(AudioBus* bus, size_t framesToProcess)
{
m_provider->provideInput(bus, framesToProcess);
}
void MediaStreamAudioSourceNode::process(size_t numberOfFrames)
{
AudioBus* outputBus = output(0)->bus();
// Use tryLock() to avoid contention in the real-time audio thread.
// If we fail to acquire the lock then the MediaStream must be in the middle of
// a format change, so we output silence in this case.
if (!m_processLock.tryLock()) {
// We failed to acquire the lock.
outputBus->zero();
return;
}
Locker locker { AdoptLock, m_processLock };
if (!m_sourceNumberOfChannels || !m_sourceSampleRate || m_sourceNumberOfChannels != outputBus->numberOfChannels()) {
outputBus->zero();
return;
}
if (numberOfFrames > outputBus->length())
numberOfFrames = outputBus->length();
if (m_multiChannelResampler) {
ASSERT(m_sourceSampleRate != sampleRate());
m_multiChannelResampler->process(outputBus, numberOfFrames);
} else {
// Bypass the resampler completely if the source is at the context's sample-rate.
ASSERT(m_sourceSampleRate == sampleRate());
provideInput(outputBus, numberOfFrames);
}
}
} // namespace WebCore
#endif // ENABLE(WEB_AUDIO) && ENABLE(MEDIA_STREAM)
|
