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1 // Copyright (c) 2010 The Chromium Authors. All rights reserved. | 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 | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include "content/renderer/audio_device.h" | 5 #include "content/renderer/audio_device.h" |
6 | 6 |
| 7 #include "base/message_loop.h" |
7 #include "base/singleton.h" | 8 #include "base/singleton.h" |
8 #include "chrome/renderer/render_thread.h" | 9 #include "chrome/renderer/render_thread.h" |
9 #include "content/common/audio_messages.h" | 10 #include "content/common/audio_messages.h" |
| 11 #include "content/common/child_process.h" |
10 #include "content/common/view_messages.h" | 12 #include "content/common/view_messages.h" |
11 #include "media/audio/audio_util.h" | 13 #include "media/audio/audio_util.h" |
12 | 14 |
13 scoped_refptr<AudioMessageFilter> AudioDevice::filter_; | 15 scoped_refptr<AudioMessageFilter> AudioDevice::filter_; |
14 | 16 |
15 namespace { | 17 namespace { |
16 | 18 |
17 // AudioMessageFilterCreator is intended to be used as a singleton so we can | 19 // AudioMessageFilterCreator is intended to be used as a singleton so we can |
18 // get access to a shared AudioMessageFilter. | 20 // get access to a shared AudioMessageFilter. |
19 // Example usage: | 21 // Example usage: |
(...skipping 14 matching lines...) Expand all Loading... |
34 } | 36 } |
35 | 37 |
36 static AudioMessageFilterCreator* GetInstance() { | 38 static AudioMessageFilterCreator* GetInstance() { |
37 return Singleton<AudioMessageFilterCreator>::get(); | 39 return Singleton<AudioMessageFilterCreator>::get(); |
38 } | 40 } |
39 | 41 |
40 private: | 42 private: |
41 scoped_refptr<AudioMessageFilter> filter_; | 43 scoped_refptr<AudioMessageFilter> filter_; |
42 }; | 44 }; |
43 | 45 |
44 } | 46 } // namespace |
45 | 47 |
46 AudioDevice::AudioDevice(size_t buffer_size, | 48 AudioDevice::AudioDevice(size_t buffer_size, |
47 int channels, | 49 int channels, |
48 double sample_rate, | 50 double sample_rate, |
49 RenderCallback* callback) | 51 RenderCallback* callback) |
50 : buffer_size_(buffer_size), | 52 : buffer_size_(buffer_size), |
51 channels_(channels), | 53 channels_(channels), |
| 54 bits_per_sample_(16), |
52 sample_rate_(sample_rate), | 55 sample_rate_(sample_rate), |
53 callback_(callback), | 56 callback_(callback), |
| 57 audio_delay_milliseconds_(0), |
| 58 volume_(1.0), |
54 stream_id_(0) { | 59 stream_id_(0) { |
55 audio_data_.reserve(channels); | 60 audio_data_.reserve(channels); |
56 for (int i = 0; i < channels; ++i) { | 61 for (int i = 0; i < channels; ++i) { |
57 float* channel_data = new float[buffer_size]; | 62 float* channel_data = new float[buffer_size]; |
58 audio_data_.push_back(channel_data); | 63 audio_data_.push_back(channel_data); |
59 } | 64 } |
| 65 // Lazily create the message filter and share across AudioDevice instances. |
| 66 filter_ = AudioMessageFilterCreator::SharedFilter(); |
60 } | 67 } |
61 | 68 |
62 AudioDevice::~AudioDevice() { | 69 AudioDevice::~AudioDevice() { |
| 70 // Make sure we have been shut down. |
| 71 DCHECK_EQ(0, stream_id_); |
63 Stop(); | 72 Stop(); |
64 for (int i = 0; i < channels_; ++i) | 73 for (int i = 0; i < channels_; ++i) |
65 delete [] audio_data_[i]; | 74 delete [] audio_data_[i]; |
66 } | 75 } |
67 | 76 |
68 bool AudioDevice::Start() { | 77 bool AudioDevice::Start() { |
69 // Make sure we don't call Start() more than once. | 78 // Make sure we don't call Start() more than once. |
70 DCHECK_EQ(0, stream_id_); | 79 DCHECK_EQ(0, stream_id_); |
71 if (stream_id_) | 80 if (stream_id_) |
72 return false; | 81 return false; |
73 | 82 |
74 // Lazily create the message filter and share across AudioDevice instances. | |
75 filter_ = AudioMessageFilterCreator::SharedFilter(); | |
76 | |
77 stream_id_ = filter_->AddDelegate(this); | |
78 | |
79 AudioParameters params; | 83 AudioParameters params; |
80 params.format = AudioParameters::AUDIO_PCM_LOW_LATENCY; | 84 params.format = AudioParameters::AUDIO_PCM_LOW_LATENCY; |
81 params.channels = channels_; | 85 params.channels = channels_; |
82 params.sample_rate = static_cast<int>(sample_rate_); | 86 params.sample_rate = static_cast<int>(sample_rate_); |
83 params.bits_per_sample = 16; | 87 params.bits_per_sample = bits_per_sample_; |
84 params.samples_per_packet = buffer_size_; | 88 params.samples_per_packet = buffer_size_; |
85 | 89 |
86 filter_->Send(new AudioHostMsg_CreateStream(0, stream_id_, params, true)); | 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)); |
87 | 101 |
88 return true; | 102 return true; |
89 } | 103 } |
90 | 104 |
91 bool AudioDevice::Stop() { | 105 bool AudioDevice::Stop() { |
92 if (stream_id_) { | 106 if (!stream_id_) |
93 OnDestroy(); | 107 return false; |
94 return true; | |
95 } | |
96 return false; | |
97 } | |
98 | 108 |
99 void AudioDevice::OnDestroy() { | 109 filter_->message_loop()->PostTask(FROM_HERE, |
100 // Make sure we don't call destroy more than once. | 110 NewRunnableMethod(this, &AudioDevice::ShutDownOnIOThread)); |
101 DCHECK_NE(0, stream_id_); | |
102 if (!stream_id_) | |
103 return; | |
104 | 111 |
105 filter_->RemoveDelegate(stream_id_); | |
106 filter_->Send(new AudioHostMsg_CloseStream(0, stream_id_)); | |
107 stream_id_ = 0; | |
108 if (audio_thread_.get()) { | 112 if (audio_thread_.get()) { |
109 socket_->Close(); | 113 socket_->Close(); |
110 audio_thread_->Join(); | 114 audio_thread_->Join(); |
111 } | 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)); |
112 } | 168 } |
113 | 169 |
114 void AudioDevice::OnRequestPacket(AudioBuffersState buffers_state) { | 170 void AudioDevice::OnRequestPacket(AudioBuffersState buffers_state) { |
115 // This method does not apply to the low-latency system. | 171 // This method does not apply to the low-latency system. |
116 NOTIMPLEMENTED(); | 172 NOTIMPLEMENTED(); |
117 } | 173 } |
118 | 174 |
119 void AudioDevice::OnStateChanged(AudioStreamState state) { | 175 void AudioDevice::OnStateChanged(AudioStreamState state) { |
120 // Not needed in this simple implementation. | 176 if (state == kAudioStreamError) { |
| 177 DLOG(WARNING) << "AudioDevice::OnStateChanged(kError)"; |
| 178 } |
121 NOTIMPLEMENTED(); | 179 NOTIMPLEMENTED(); |
122 } | 180 } |
123 | 181 |
124 void AudioDevice::OnCreated( | 182 void AudioDevice::OnCreated( |
125 base::SharedMemoryHandle handle, uint32 length) { | 183 base::SharedMemoryHandle handle, uint32 length) { |
126 // Not needed in this simple implementation. | 184 // Not needed in this simple implementation. |
127 NOTIMPLEMENTED(); | 185 NOTIMPLEMENTED(); |
128 } | 186 } |
129 | 187 |
130 void AudioDevice::OnLowLatencyCreated( | 188 void AudioDevice::OnLowLatencyCreated( |
131 base::SharedMemoryHandle handle, | 189 base::SharedMemoryHandle handle, |
132 base::SyncSocket::Handle socket_handle, | 190 base::SyncSocket::Handle socket_handle, |
133 uint32 length) { | 191 uint32 length) { |
134 | |
135 #if defined(OS_WIN) | 192 #if defined(OS_WIN) |
136 DCHECK(handle); | 193 DCHECK(handle); |
137 DCHECK(socket_handle); | 194 DCHECK(socket_handle); |
138 #else | 195 #else |
139 DCHECK_GE(handle.fd, 0); | 196 DCHECK_GE(handle.fd, 0); |
140 DCHECK_GE(socket_handle, 0); | 197 DCHECK_GE(socket_handle, 0); |
141 #endif | 198 #endif |
142 DCHECK(length); | 199 DCHECK(length); |
143 DCHECK(!audio_thread_.get()); | |
144 | 200 |
145 // TODO(crogers) : check that length is big enough for buffer_size_ | 201 // TODO(crogers) : check that length is big enough for buffer_size_ |
146 | 202 |
147 shared_memory_.reset(new base::SharedMemory(handle, false)); | 203 shared_memory_.reset(new base::SharedMemory(handle, false)); |
148 shared_memory_->Map(length); | 204 shared_memory_->Map(length); |
149 | 205 |
150 socket_.reset(new base::SyncSocket(socket_handle)); | 206 socket_.reset(new base::SyncSocket(socket_handle)); |
151 // Allow the client to pre-populate the buffer. | 207 // Allow the client to pre-populate the buffer. |
152 FireRenderCallback(); | 208 FireRenderCallback(); |
153 | 209 |
154 // TODO(crogers): we could optionally set the thread to high-priority | 210 // TODO(crogers): we could optionally set the thread to high-priority |
155 audio_thread_.reset( | 211 audio_thread_.reset( |
156 new base::DelegateSimpleThread(this, "renderer_audio_thread")); | 212 new base::DelegateSimpleThread(this, "renderer_audio_thread")); |
157 audio_thread_->Start(); | 213 audio_thread_->Start(); |
158 | 214 |
159 filter_->Send(new AudioHostMsg_PlayStream(0, stream_id_)); | 215 if (filter_) { |
| 216 filter_->message_loop()->PostTask(FROM_HERE, |
| 217 NewRunnableMethod(this, &AudioDevice::StartOnIOThread)); |
| 218 } |
160 } | 219 } |
161 | 220 |
162 void AudioDevice::OnVolume(double volume) { | 221 void AudioDevice::OnVolume(double volume) { |
163 // Not needed in this simple implementation. | |
164 NOTIMPLEMENTED(); | 222 NOTIMPLEMENTED(); |
165 } | 223 } |
166 | 224 |
167 // Our audio thread runs here. | 225 // Our audio thread runs here. |
168 void AudioDevice::Run() { | 226 void AudioDevice::Run() { |
169 int pending_data; | 227 int pending_data; |
| 228 const int samples_per_ms = static_cast<int>(sample_rate_) / 1000; |
| 229 const int bytes_per_ms = channels_ * (bits_per_sample_ / 8) * samples_per_ms; |
| 230 |
170 while (sizeof(pending_data) == socket_->Receive(&pending_data, | 231 while (sizeof(pending_data) == socket_->Receive(&pending_data, |
171 sizeof(pending_data)) && | 232 sizeof(pending_data)) && |
172 pending_data >= 0) { | 233 pending_data >= 0) { |
| 234 { |
| 235 // Convert the number of pending bytes in the render buffer |
| 236 // into milliseconds. |
| 237 audio_delay_milliseconds_ = pending_data / bytes_per_ms; |
| 238 } |
| 239 |
173 FireRenderCallback(); | 240 FireRenderCallback(); |
174 } | 241 } |
175 } | 242 } |
176 | 243 |
177 void AudioDevice::FireRenderCallback() { | 244 void AudioDevice::FireRenderCallback() { |
178 if (callback_) { | 245 if (callback_) { |
179 // Ask client to render audio. | 246 // Update the audio-delay measurement then ask client to render audio. |
180 callback_->Render(audio_data_, buffer_size_); | 247 callback_->Render(audio_data_, buffer_size_, audio_delay_milliseconds_); |
181 | 248 |
182 // Interleave, scale, and clip to int16. | 249 // Interleave, scale, and clip to int16. |
183 int16* output_buffer16 = static_cast<int16*>(shared_memory_data()); | 250 int16* output_buffer16 = static_cast<int16*>(shared_memory_data()); |
184 media::InterleaveFloatToInt16(audio_data_, output_buffer16, buffer_size_); | 251 media::InterleaveFloatToInt16(audio_data_, output_buffer16, buffer_size_); |
185 } | 252 } |
186 } | 253 } |
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