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| 1 // Copyright (c) 2010 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 "chrome/renderer/audio_device.h" | |
| 6 | |
| 7 #include "base/singleton.h" | |
| 8 #include "chrome/common/render_messages.h" | |
| 9 #include "chrome/common/render_messages_params.h" | |
| 10 #include "chrome/renderer/render_thread.h" | |
| 11 #include "media/audio/audio_util.h" | |
| 12 | |
| 13 scoped_refptr<AudioMessageFilter> AudioDevice::filter_; | |
| 14 | |
| 15 namespace { | |
| 16 | |
| 17 // AudioMessageFilterCreator is intended to be used as a singleton so we can | |
| 18 // get access to a shared AudioMessageFilter. | |
| 19 // Example usage: | |
| 20 // AudioMessageFilter* filter = AudioMessageFilterCreator::SharedFilter(); | |
| 21 | |
| 22 class AudioMessageFilterCreator { | |
| 23 public: | |
| 24 AudioMessageFilterCreator() { | |
| 25 int routing_id; | |
| 26 RenderThread::current()->Send( | |
| 27 new ViewHostMsg_GenerateRoutingID(&routing_id)); | |
| 28 filter_ = new AudioMessageFilter(routing_id); | |
| 29 RenderThread::current()->AddFilter(filter_); | |
| 30 } | |
| 31 | |
| 32 static AudioMessageFilter* SharedFilter() { | |
| 33 return GetInstance()->filter_.get(); | |
| 34 } | |
| 35 | |
| 36 static AudioMessageFilterCreator* GetInstance() { | |
| 37 return Singleton<AudioMessageFilterCreator>::get(); | |
| 38 } | |
| 39 | |
| 40 private: | |
| 41 scoped_refptr<AudioMessageFilter> filter_; | |
| 42 }; | |
| 43 | |
| 44 } | |
| 45 | |
| 46 AudioDevice::AudioDevice(size_t buffer_size, | |
| 47 int channels, | |
| 48 double sample_rate, | |
| 49 RenderCallback* callback) | |
| 50 : buffer_size_(buffer_size), | |
| 51 channels_(channels), | |
| 52 sample_rate_(sample_rate), | |
| 53 callback_(callback), | |
| 54 stream_id_(0) { | |
| 55 audio_data_.reserve(channels); | |
| 56 for (int i = 0; i < channels; ++i) { | |
| 57 float* channel_data = new float[buffer_size]; | |
| 58 audio_data_.push_back(channel_data); | |
| 59 } | |
| 60 } | |
| 61 | |
| 62 AudioDevice::~AudioDevice() { | |
| 63 Stop(); | |
| 64 for (int i = 0; i < channels_; ++i) | |
| 65 delete [] audio_data_[i]; | |
| 66 } | |
| 67 | |
| 68 bool AudioDevice::Start() { | |
| 69 // Make sure we don't call Start() more than once. | |
| 70 DCHECK_EQ(0, stream_id_); | |
| 71 if (stream_id_) | |
| 72 return false; | |
| 73 | |
| 74 // Lazily create the message filter and share across AudioDevice instances. | |
| 75 filter_ = AudioMessageFilterCreator::SharedFilter(); | |
| 76 | |
| 77 stream_id_ = filter_->AddDelegate(this); | |
| 78 | |
| 79 ViewHostMsg_Audio_CreateStream_Params params; | |
| 80 params.params.format = AudioParameters::AUDIO_PCM_LOW_LATENCY; | |
| 81 params.params.channels = channels_; | |
| 82 params.params.sample_rate = static_cast<int>(sample_rate_); | |
| 83 params.params.bits_per_sample = 16; | |
| 84 params.params.samples_per_packet = buffer_size_; | |
| 85 | |
| 86 filter_->Send( | |
| 87 new ViewHostMsg_CreateAudioStream(0, stream_id_, params, true)); | |
| 88 | |
| 89 return true; | |
| 90 } | |
| 91 | |
| 92 bool AudioDevice::Stop() { | |
| 93 if (stream_id_) { | |
| 94 OnDestroy(); | |
| 95 return true; | |
| 96 } | |
| 97 return false; | |
| 98 } | |
| 99 | |
| 100 void AudioDevice::OnDestroy() { | |
| 101 // Make sure we don't call destroy more than once. | |
| 102 DCHECK_NE(0, stream_id_); | |
| 103 if (!stream_id_) | |
| 104 return; | |
| 105 | |
| 106 filter_->RemoveDelegate(stream_id_); | |
| 107 filter_->Send(new ViewHostMsg_CloseAudioStream(0, stream_id_)); | |
| 108 stream_id_ = 0; | |
| 109 if (audio_thread_.get()) { | |
| 110 socket_->Close(); | |
| 111 audio_thread_->Join(); | |
| 112 } | |
| 113 } | |
| 114 | |
| 115 void AudioDevice::OnRequestPacket(AudioBuffersState buffers_state) { | |
| 116 // This method does not apply to the low-latency system. | |
| 117 NOTIMPLEMENTED(); | |
| 118 } | |
| 119 | |
| 120 void AudioDevice::OnStateChanged( | |
| 121 const ViewMsg_AudioStreamState_Params& state) { | |
| 122 // Not needed in this simple implementation. | |
| 123 NOTIMPLEMENTED(); | |
| 124 } | |
| 125 | |
| 126 void AudioDevice::OnCreated( | |
| 127 base::SharedMemoryHandle handle, uint32 length) { | |
| 128 // Not needed in this simple implementation. | |
| 129 NOTIMPLEMENTED(); | |
| 130 } | |
| 131 | |
| 132 void AudioDevice::OnLowLatencyCreated( | |
| 133 base::SharedMemoryHandle handle, | |
| 134 base::SyncSocket::Handle socket_handle, | |
| 135 uint32 length) { | |
| 136 | |
| 137 #if defined(OS_WIN) | |
| 138 DCHECK(handle); | |
| 139 DCHECK(socket_handle); | |
| 140 #else | |
| 141 DCHECK_GE(handle.fd, 0); | |
| 142 DCHECK_GE(socket_handle, 0); | |
| 143 #endif | |
| 144 DCHECK(length); | |
| 145 DCHECK(!audio_thread_.get()); | |
| 146 | |
| 147 // TODO(crogers) : check that length is big enough for buffer_size_ | |
| 148 | |
| 149 shared_memory_.reset(new base::SharedMemory(handle, false)); | |
| 150 shared_memory_->Map(length); | |
| 151 | |
| 152 socket_.reset(new base::SyncSocket(socket_handle)); | |
| 153 // Allow the client to pre-populate the buffer. | |
| 154 FireRenderCallback(); | |
| 155 | |
| 156 // TODO(crogers): we could optionally set the thread to high-priority | |
| 157 audio_thread_.reset( | |
| 158 new base::DelegateSimpleThread(this, "renderer_audio_thread")); | |
| 159 audio_thread_->Start(); | |
| 160 | |
| 161 filter_->Send(new ViewHostMsg_PlayAudioStream(0, stream_id_)); | |
| 162 } | |
| 163 | |
| 164 void AudioDevice::OnVolume(double volume) { | |
| 165 // Not needed in this simple implementation. | |
| 166 NOTIMPLEMENTED(); | |
| 167 } | |
| 168 | |
| 169 // Our audio thread runs here. | |
| 170 void AudioDevice::Run() { | |
| 171 int pending_data; | |
| 172 while (sizeof(pending_data) == socket_->Receive(&pending_data, | |
| 173 sizeof(pending_data)) && | |
| 174 pending_data >= 0) { | |
| 175 FireRenderCallback(); | |
| 176 } | |
| 177 } | |
| 178 | |
| 179 void AudioDevice::FireRenderCallback() { | |
| 180 if (callback_) { | |
| 181 // Ask client to render audio. | |
| 182 callback_->Render(audio_data_, buffer_size_); | |
| 183 | |
| 184 // Interleave, scale, and clip to int16. | |
| 185 int16* output_buffer16 = static_cast<int16*>(shared_memory_data()); | |
| 186 media::InterleaveFloatToInt16(audio_data_, output_buffer16, buffer_size_); | |
| 187 } | |
| 188 } | |
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