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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2012 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/media/render_audiosourceprovider.h" | 5 #include "content/renderer/media/render_audiosourceprovider.h" |
6 | 6 |
7 #include "base/basictypes.h" | 7 #include "base/basictypes.h" |
8 #include "base/logging.h" | 8 #include "base/logging.h" |
9 #include "third_party/WebKit/Source/WebKit/chromium/public/WebAudioSourceProvide
rClient.h" | 9 #include "third_party/WebKit/Source/WebKit/chromium/public/WebAudioSourceProvide
rClient.h" |
10 | 10 |
11 using std::vector; | 11 using std::vector; |
12 using WebKit::WebVector; | 12 using WebKit::WebVector; |
13 | 13 |
14 RenderAudioSourceProvider::RenderAudioSourceProvider() | 14 RenderAudioSourceProvider::RenderAudioSourceProvider() |
15 : is_initialized_(false), | 15 : is_initialized_(false), |
16 channels_(0), | 16 channels_(0), |
17 sample_rate_(0.0), | 17 sample_rate_(0), |
18 is_running_(false), | 18 is_running_(false), |
19 volume_(1.0), | 19 volume_(1.0), |
20 renderer_(NULL), | 20 renderer_(NULL), |
21 client_(NULL) { | 21 client_(NULL) { |
22 // We create the AudioDevice here because it must be created in the | 22 // We create the AudioDevice here because it must be created in the |
23 // main thread. But we don't yet know the audio format (sample-rate, etc.) | 23 // main thread. But we don't yet know the audio format (sample-rate, etc.) |
24 // at this point. Later, when Initialize() is called, we have | 24 // at this point. Later, when Initialize() is called, we have |
25 // the audio format information and call the AudioDevice::Initialize() | 25 // the audio format information and call the AudioDevice::Initialize() |
26 // method to fully initialize it. | 26 // method to fully initialize it. |
27 default_sink_ = new AudioDevice(); | 27 default_sink_ = new AudioDevice(); |
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66 } | 66 } |
67 | 67 |
68 void RenderAudioSourceProvider::GetVolume(double* volume) { | 68 void RenderAudioSourceProvider::GetVolume(double* volume) { |
69 if (!client_) | 69 if (!client_) |
70 default_sink_->GetVolume(volume); | 70 default_sink_->GetVolume(volume); |
71 else if (volume) | 71 else if (volume) |
72 *volume = volume_; | 72 *volume = volume_; |
73 } | 73 } |
74 | 74 |
75 void RenderAudioSourceProvider::Initialize( | 75 void RenderAudioSourceProvider::Initialize( |
76 size_t buffer_size, | 76 const AudioParameters& params, RenderCallback* renderer) { |
77 int channels, | |
78 double sample_rate, | |
79 AudioParameters::Format latency_format, | |
80 RenderCallback* renderer) { | |
81 base::AutoLock auto_lock(sink_lock_); | 77 base::AutoLock auto_lock(sink_lock_); |
82 CHECK(!is_initialized_); | 78 CHECK(!is_initialized_); |
83 renderer_ = renderer; | 79 renderer_ = renderer; |
84 | 80 |
85 default_sink_->Initialize(buffer_size, | 81 default_sink_->Initialize(params, renderer); |
86 channels, | 82 |
87 sample_rate, | 83 // Keep track of the format in case the client hasn't yet been set. |
88 latency_format, | 84 channels_ = params.channels(); |
89 renderer); | 85 sample_rate_ = params.sample_rate(); |
90 | 86 |
91 if (client_) { | 87 if (client_) { |
92 // Inform WebKit about the audio stream format. | 88 // Inform WebKit about the audio stream format. |
93 client_->setFormat(channels, sample_rate); | 89 client_->setFormat(channels_, sample_rate_); |
94 } | 90 } |
95 | 91 |
96 // Keep track of the format in case the client hasn't yet been set. | |
97 channels_ = channels; | |
98 sample_rate_ = sample_rate; | |
99 is_initialized_ = true; | 92 is_initialized_ = true; |
100 } | 93 } |
101 | 94 |
102 void RenderAudioSourceProvider::setClient( | 95 void RenderAudioSourceProvider::setClient( |
103 WebKit::WebAudioSourceProviderClient* client) { | 96 WebKit::WebAudioSourceProviderClient* client) { |
104 // Synchronize with other uses of client_ and default_sink_. | 97 // Synchronize with other uses of client_ and default_sink_. |
105 base::AutoLock auto_lock(sink_lock_); | 98 base::AutoLock auto_lock(sink_lock_); |
106 | 99 |
107 if (client && client != client_) { | 100 if (client && client != client_) { |
108 // Detach the audio renderer from normal playback. | 101 // Detach the audio renderer from normal playback. |
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139 | 132 |
140 // TODO(crogers): figure out if we should volume scale here or in common | 133 // TODO(crogers): figure out if we should volume scale here or in common |
141 // WebAudio code. In any case we need to take care of volume. | 134 // WebAudio code. In any case we need to take care of volume. |
142 renderer_->Render(v, number_of_frames, 0); | 135 renderer_->Render(v, number_of_frames, 0); |
143 } else { | 136 } else { |
144 // Provide silence if the source is not running. | 137 // Provide silence if the source is not running. |
145 for (size_t i = 0; i < audio_data.size(); ++i) | 138 for (size_t i = 0; i < audio_data.size(); ++i) |
146 memset(audio_data[i], 0, sizeof(float) * number_of_frames); | 139 memset(audio_data[i], 0, sizeof(float) * number_of_frames); |
147 } | 140 } |
148 } | 141 } |
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