1 /*
2 * Copyright (C) 2010 Google Inc. All rights reserved.
3 *
4 * Redistribution and use in source and binary forms, with or without
5 * modification, are permitted provided that the following conditions
6 * are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright
9 * notice, this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright
11 * notice, this list of conditions and the following disclaimer in the
12 * documentation and/or other materials provided with the distribution.
13 * 3. Neither the name of Apple Computer, Inc. ("Apple") nor the names of
14 * its contributors may be used to endorse or promote products derived
15 * from this software without specific prior written permission.
16 *
17 * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
18 * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
19 * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
20 * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
21 * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
22 * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
23 * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
24 * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
25 * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
26 * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
27 */
28
29 #include "config.h"
30
31 #if ENABLE(WEB_AUDIO)
32
33 #include "platform/audio/AudioDestination.h"
34
35 #include "platform/audio/AudioFIFO.h"
36 #include "platform/audio/AudioPullFIFO.h"
37 #include "public/platform/Platform.h"
38
39 namespace WebCore {
40
41 // Buffer size at which the web audio engine will render.
42 const unsigned renderBufferSize = 128;
43
44 // Size of the FIFO
45 const size_t fifoSize = 8192;
46
47 // Factory method: Chromium-implementation
create(AudioIOCallback & callback,const String & inputDeviceId,unsigned numberOfInputChannels,unsigned numberOfOutputChannels,float sampleRate)48 PassOwnPtr<AudioDestination> AudioDestination::create(AudioIOCallback& callback, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate)
49 {
50 return adoptPtr(new AudioDestination(callback, inputDeviceId, numberOfInputChannels, numberOfOutputChannels, sampleRate));
51 }
52
AudioDestination(AudioIOCallback & callback,const String & inputDeviceId,unsigned numberOfInputChannels,unsigned numberOfOutputChannels,float sampleRate)53 AudioDestination::AudioDestination(AudioIOCallback& callback, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate)
54 : m_callback(callback)
55 , m_numberOfOutputChannels(numberOfOutputChannels)
56 , m_inputBus(AudioBus::create(numberOfInputChannels, renderBufferSize))
57 , m_renderBus(AudioBus::create(numberOfOutputChannels, renderBufferSize, false))
58 , m_sampleRate(sampleRate)
59 , m_isPlaying(false)
60 {
61 // Use the optimal buffer size recommended by the audio backend.
62 m_callbackBufferSize = blink::Platform::current()->audioHardwareBufferSize();
63
64 #if OS(ANDROID)
65 // The optimum low-latency hardware buffer size is usually too small on Android for WebAudio to
66 // render without glitching. So, if it is small, use a larger size. If it was already large, use
67 // the requested size.
68 //
69 // Since WebAudio renders in 128-frame blocks, the small buffer sizes (144 for a Galaxy Nexus),
70 // cause significant processing jitter. Sometimes multiple blocks will processed, but other
71 // times will not be since the FIFO can satisfy the request. By using a larger
72 // callbackBufferSize, we smooth out the jitter.
73 const size_t kSmallBufferSize = 1024;
74 const size_t kDefaultCallbackBufferSize = 2048;
75
76 if (m_callbackBufferSize <= kSmallBufferSize)
77 m_callbackBufferSize = kDefaultCallbackBufferSize;
78 #endif
79
80 // Quick exit if the requested size is too large.
81 ASSERT(m_callbackBufferSize + renderBufferSize <= fifoSize);
82 if (m_callbackBufferSize + renderBufferSize > fifoSize)
83 return;
84
85 m_audioDevice = adoptPtr(blink::Platform::current()->createAudioDevice(m_callbackBufferSize, numberOfInputChannels, numberOfOutputChannels, sampleRate, this, inputDeviceId));
86 ASSERT(m_audioDevice);
87
88 // Create a FIFO to handle the possibility of the callback size
89 // not being a multiple of the render size. If the FIFO already
90 // contains enough data, the data will be provided directly.
91 // Otherwise, the FIFO will call the provider enough times to
92 // satisfy the request for data.
93 m_fifo = adoptPtr(new AudioPullFIFO(*this, numberOfOutputChannels, fifoSize, renderBufferSize));
94
95 // Input buffering.
96 m_inputFifo = adoptPtr(new AudioFIFO(numberOfInputChannels, fifoSize));
97
98 // If the callback size does not match the render size, then we need to buffer some
99 // extra silence for the input. Otherwise, we can over-consume the input FIFO.
100 if (m_callbackBufferSize != renderBufferSize) {
101 // FIXME: handle multi-channel input and don't hard-code to stereo.
102 RefPtr<AudioBus> silence = AudioBus::create(2, renderBufferSize);
103 m_inputFifo->push(silence.get());
104 }
105 }
106
~AudioDestination()107 AudioDestination::~AudioDestination()
108 {
109 stop();
110 }
111
start()112 void AudioDestination::start()
113 {
114 if (!m_isPlaying && m_audioDevice) {
115 m_audioDevice->start();
116 m_isPlaying = true;
117 }
118 }
119
stop()120 void AudioDestination::stop()
121 {
122 if (m_isPlaying && m_audioDevice) {
123 m_audioDevice->stop();
124 m_isPlaying = false;
125 }
126 }
127
hardwareSampleRate()128 float AudioDestination::hardwareSampleRate()
129 {
130 return static_cast<float>(blink::Platform::current()->audioHardwareSampleRate());
131 }
132
maxChannelCount()133 unsigned long AudioDestination::maxChannelCount()
134 {
135 return static_cast<float>(blink::Platform::current()->audioHardwareOutputChannels());
136 }
137
render(const blink::WebVector<float * > & sourceData,const blink::WebVector<float * > & audioData,size_t numberOfFrames)138 void AudioDestination::render(const blink::WebVector<float*>& sourceData, const blink::WebVector<float*>& audioData, size_t numberOfFrames)
139 {
140 bool isNumberOfChannelsGood = audioData.size() == m_numberOfOutputChannels;
141 if (!isNumberOfChannelsGood) {
142 ASSERT_NOT_REACHED();
143 return;
144 }
145
146 bool isBufferSizeGood = numberOfFrames == m_callbackBufferSize;
147 if (!isBufferSizeGood) {
148 ASSERT_NOT_REACHED();
149 return;
150 }
151
152 // Buffer optional live input.
153 if (sourceData.size() >= 2) {
154 // FIXME: handle multi-channel input and don't hard-code to stereo.
155 RefPtr<AudioBus> wrapperBus = AudioBus::create(2, numberOfFrames, false);
156 wrapperBus->setChannelMemory(0, sourceData[0], numberOfFrames);
157 wrapperBus->setChannelMemory(1, sourceData[1], numberOfFrames);
158 m_inputFifo->push(wrapperBus.get());
159 }
160
161 for (unsigned i = 0; i < m_numberOfOutputChannels; ++i)
162 m_renderBus->setChannelMemory(i, audioData[i], numberOfFrames);
163
164 m_fifo->consume(m_renderBus.get(), numberOfFrames);
165 }
166
provideInput(AudioBus * bus,size_t framesToProcess)167 void AudioDestination::provideInput(AudioBus* bus, size_t framesToProcess)
168 {
169 AudioBus* sourceBus = 0;
170 if (m_inputFifo->framesInFifo() >= framesToProcess) {
171 m_inputFifo->consume(m_inputBus.get(), framesToProcess);
172 sourceBus = m_inputBus.get();
173 }
174
175 m_callback.render(sourceBus, bus, framesToProcess);
176 }
177
178 } // namespace WebCore
179
180 #endif // ENABLE(WEB_AUDIO)
181