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1 // Copyright (c) 2011 The Chromium Authors. All rights reserved. | |
2 // Use of this source code is governed by a BSD-style license that can be | |
3 // found in the LICENSE file. | |
4 | |
5 #include "content/renderer/audio_device.h" | |
6 | |
7 #include "base/memory/singleton.h" | |
8 #include "base/message_loop.h" | |
9 #include "content/common/audio_messages.h" | |
10 #include "content/common/child_process.h" | |
11 #include "content/common/view_messages.h" | |
12 #include "content/renderer/render_thread.h" | |
13 #include "media/audio/audio_util.h" | |
14 | |
15 scoped_refptr<AudioMessageFilter> AudioDevice::filter_; | |
16 | |
17 namespace { | |
18 | |
19 // AudioMessageFilterCreator is intended to be used as a singleton so we can | |
20 // get access to a shared AudioMessageFilter. | |
21 // Example usage: | |
22 // AudioMessageFilter* filter = AudioMessageFilterCreator::SharedFilter(); | |
23 | |
24 class AudioMessageFilterCreator { | |
25 public: | |
26 AudioMessageFilterCreator() { | |
27 int routing_id; | |
28 RenderThread::current()->Send( | |
29 new ViewHostMsg_GenerateRoutingID(&routing_id)); | |
30 filter_ = new AudioMessageFilter(routing_id); | |
31 RenderThread::current()->AddFilter(filter_); | |
32 } | |
33 | |
34 static AudioMessageFilter* SharedFilter() { | |
35 return GetInstance()->filter_.get(); | |
36 } | |
37 | |
38 static AudioMessageFilterCreator* GetInstance() { | |
39 return Singleton<AudioMessageFilterCreator>::get(); | |
40 } | |
41 | |
42 private: | |
43 scoped_refptr<AudioMessageFilter> filter_; | |
44 }; | |
45 | |
46 } // namespace | |
47 | |
48 AudioDevice::AudioDevice(size_t buffer_size, | |
49 int channels, | |
50 double sample_rate, | |
51 RenderCallback* callback) | |
52 : buffer_size_(buffer_size), | |
53 channels_(channels), | |
54 bits_per_sample_(16), | |
55 sample_rate_(sample_rate), | |
56 callback_(callback), | |
57 audio_delay_milliseconds_(0), | |
58 volume_(1.0), | |
59 stream_id_(0) { | |
60 audio_data_.reserve(channels); | |
61 for (int i = 0; i < channels; ++i) { | |
62 float* channel_data = new float[buffer_size]; | |
63 audio_data_.push_back(channel_data); | |
64 } | |
65 // Lazily create the message filter and share across AudioDevice instances. | |
66 filter_ = AudioMessageFilterCreator::SharedFilter(); | |
67 } | |
68 | |
69 AudioDevice::~AudioDevice() { | |
70 // Make sure we have been shut down. | |
71 DCHECK_EQ(0, stream_id_); | |
72 Stop(); | |
73 for (int i = 0; i < channels_; ++i) | |
74 delete [] audio_data_[i]; | |
75 } | |
76 | |
77 bool AudioDevice::Start() { | |
78 // Make sure we don't call Start() more than once. | |
79 DCHECK_EQ(0, stream_id_); | |
80 if (stream_id_) | |
81 return false; | |
82 | |
83 AudioParameters params; | |
84 params.format = AudioParameters::AUDIO_PCM_LOW_LATENCY; | |
85 params.channels = channels_; | |
86 params.sample_rate = static_cast<int>(sample_rate_); | |
87 params.bits_per_sample = bits_per_sample_; | |
88 params.samples_per_packet = buffer_size_; | |
89 | |
90 // Ensure that the initialization task is posted on the I/O thread by | |
91 // accessing the I/O message loop directly. This approach avoids a race | |
92 // condition which could exist if the message loop of the filter was | |
93 // used instead. | |
94 DCHECK(ChildProcess::current()) << "Must be in the renderer"; | |
95 MessageLoop* message_loop = ChildProcess::current()->io_message_loop(); | |
96 if (!message_loop) | |
97 return false; | |
98 | |
99 message_loop->PostTask(FROM_HERE, | |
100 NewRunnableMethod(this, &AudioDevice::InitializeOnIOThread, params)); | |
101 | |
102 return true; | |
103 } | |
104 | |
105 bool AudioDevice::Stop() { | |
106 if (!stream_id_) | |
107 return false; | |
108 | |
109 filter_->message_loop()->PostTask(FROM_HERE, | |
110 NewRunnableMethod(this, &AudioDevice::ShutDownOnIOThread)); | |
111 | |
112 if (audio_thread_.get()) { | |
113 socket_->Close(); | |
114 audio_thread_->Join(); | |
115 } | |
116 | |
117 return true; | |
118 } | |
119 | |
120 bool AudioDevice::SetVolume(double volume) { | |
121 if (!stream_id_) | |
122 return false; | |
123 | |
124 if (volume < 0 || volume > 1.0) | |
125 return false; | |
126 | |
127 filter_->message_loop()->PostTask(FROM_HERE, | |
128 NewRunnableMethod(this, &AudioDevice::SetVolumeOnIOThread, volume)); | |
129 | |
130 volume_ = volume; | |
131 | |
132 return true; | |
133 } | |
134 | |
135 bool AudioDevice::GetVolume(double* volume) { | |
136 if (!stream_id_) | |
137 return false; | |
138 | |
139 // Return a locally cached version of the current scaling factor. | |
140 *volume = volume_; | |
141 | |
142 return true; | |
143 } | |
144 | |
145 void AudioDevice::InitializeOnIOThread(const AudioParameters& params) { | |
146 stream_id_ = filter_->AddDelegate(this); | |
147 filter_->Send(new AudioHostMsg_CreateStream(0, stream_id_, params, true)); | |
148 } | |
149 | |
150 void AudioDevice::StartOnIOThread() { | |
151 if (stream_id_) | |
152 filter_->Send(new AudioHostMsg_PlayStream(0, stream_id_)); | |
153 } | |
154 | |
155 void AudioDevice::ShutDownOnIOThread() { | |
156 // Make sure we don't call shutdown more than once. | |
157 if (!stream_id_) | |
158 return; | |
159 | |
160 filter_->Send(new AudioHostMsg_CloseStream(0, stream_id_)); | |
161 filter_->RemoveDelegate(stream_id_); | |
162 stream_id_ = 0; | |
163 } | |
164 | |
165 void AudioDevice::SetVolumeOnIOThread(double volume) { | |
166 if (stream_id_) | |
167 filter_->Send(new AudioHostMsg_SetVolume(0, stream_id_, volume)); | |
168 } | |
169 | |
170 void AudioDevice::OnRequestPacket(AudioBuffersState buffers_state) { | |
171 // This method does not apply to the low-latency system. | |
172 NOTIMPLEMENTED(); | |
173 } | |
174 | |
175 void AudioDevice::OnStateChanged(AudioStreamState state) { | |
176 if (state == kAudioStreamError) { | |
177 DLOG(WARNING) << "AudioDevice::OnStateChanged(kError)"; | |
178 } | |
179 NOTIMPLEMENTED(); | |
180 } | |
181 | |
182 void AudioDevice::OnCreated( | |
183 base::SharedMemoryHandle handle, uint32 length) { | |
184 // Not needed in this simple implementation. | |
185 NOTIMPLEMENTED(); | |
186 } | |
187 | |
188 void AudioDevice::OnLowLatencyCreated( | |
189 base::SharedMemoryHandle handle, | |
190 base::SyncSocket::Handle socket_handle, | |
191 uint32 length) { | |
192 #if defined(OS_WIN) | |
193 DCHECK(handle); | |
194 DCHECK(socket_handle); | |
195 #else | |
196 DCHECK_GE(handle.fd, 0); | |
197 DCHECK_GE(socket_handle, 0); | |
198 #endif | |
199 DCHECK(length); | |
200 | |
201 // TODO(crogers) : check that length is big enough for buffer_size_ | |
202 | |
203 shared_memory_.reset(new base::SharedMemory(handle, false)); | |
204 shared_memory_->Map(length); | |
205 | |
206 socket_.reset(new base::SyncSocket(socket_handle)); | |
207 // Allow the client to pre-populate the buffer. | |
208 FireRenderCallback(); | |
209 | |
210 audio_thread_.reset( | |
211 new base::DelegateSimpleThread(this, "renderer_audio_thread")); | |
212 audio_thread_->Start(); | |
213 | |
214 if (filter_) { | |
215 filter_->message_loop()->PostTask(FROM_HERE, | |
216 NewRunnableMethod(this, &AudioDevice::StartOnIOThread)); | |
217 } | |
218 } | |
219 | |
220 void AudioDevice::OnVolume(double volume) { | |
221 NOTIMPLEMENTED(); | |
222 } | |
223 | |
224 // Our audio thread runs here. | |
225 void AudioDevice::Run() { | |
226 audio_thread_->SetThreadPriority(base::kThreadPriority_RealtimeAudio); | |
227 | |
228 int pending_data; | |
229 const int samples_per_ms = static_cast<int>(sample_rate_) / 1000; | |
230 const int bytes_per_ms = channels_ * (bits_per_sample_ / 8) * samples_per_ms; | |
231 | |
232 while (sizeof(pending_data) == socket_->Receive(&pending_data, | |
233 sizeof(pending_data)) && | |
234 pending_data >= 0) { | |
235 { | |
236 // Convert the number of pending bytes in the render buffer | |
237 // into milliseconds. | |
238 audio_delay_milliseconds_ = pending_data / bytes_per_ms; | |
239 } | |
240 | |
241 FireRenderCallback(); | |
242 } | |
243 } | |
244 | |
245 void AudioDevice::FireRenderCallback() { | |
246 if (callback_) { | |
247 // Update the audio-delay measurement then ask client to render audio. | |
248 callback_->Render(audio_data_, buffer_size_, audio_delay_milliseconds_); | |
249 | |
250 // Interleave, scale, and clip to int16. | |
251 int16* output_buffer16 = static_cast<int16*>(shared_memory_data()); | |
252 media::InterleaveFloatToInt16(audio_data_, output_buffer16, buffer_size_); | |
253 } | |
254 } | |
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