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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ |
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ | 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ |
13 | 13 |
14 #include <list> | 14 #include <list> |
15 #include <string> | 15 #include <string> |
16 #include <vector> | 16 #include <vector> |
17 | 17 |
18 #include "webrtc/base/scoped_ptr.h" | 18 #include "webrtc/base/scoped_ptr.h" |
19 #include "webrtc/base/thread_annotations.h" | 19 #include "webrtc/base/thread_annotations.h" |
20 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 20 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
21 | 21 |
22 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP | |
peah-webrtc
2015/10/02 08:34:27
Is this really the correct way to do this? Why not
minyue-webrtc
2015/10/02 09:01:42
problem goes away with new change in Patch set 7
| |
23 // Files generated at build-time by the protobuf compiler. | |
24 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD | |
25 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h" | |
26 #else | |
27 #include "webrtc/audio_processing/debug.pb.h" | |
28 #endif | |
29 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP | |
30 | |
22 namespace webrtc { | 31 namespace webrtc { |
23 | 32 |
24 class AgcManagerDirect; | 33 class AgcManagerDirect; |
25 class AudioBuffer; | 34 class AudioBuffer; |
26 class AudioConverter; | 35 class AudioConverter; |
27 | 36 |
28 template<typename T> | 37 template<typename T> |
29 class Beamformer; | 38 class Beamformer; |
30 | 39 |
31 class CriticalSectionWrapper; | 40 class CriticalSectionWrapper; |
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160 std::list<ProcessingComponent*> component_list_; | 169 std::list<ProcessingComponent*> component_list_; |
161 CriticalSectionWrapper* crit_; | 170 CriticalSectionWrapper* crit_; |
162 rtc::scoped_ptr<AudioBuffer> render_audio_; | 171 rtc::scoped_ptr<AudioBuffer> render_audio_; |
163 rtc::scoped_ptr<AudioBuffer> capture_audio_; | 172 rtc::scoped_ptr<AudioBuffer> capture_audio_; |
164 rtc::scoped_ptr<AudioConverter> render_converter_; | 173 rtc::scoped_ptr<AudioConverter> render_converter_; |
165 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP | 174 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP |
166 // TODO(andrew): make this more graceful. Ideally we would split this stuff | 175 // TODO(andrew): make this more graceful. Ideally we would split this stuff |
167 // out into a separate class with an "enabled" and "disabled" implementation. | 176 // out into a separate class with an "enabled" and "disabled" implementation. |
168 int WriteMessageToDebugFile(); | 177 int WriteMessageToDebugFile(); |
169 int WriteInitMessage(); | 178 int WriteInitMessage(); |
179 int WriteConfigMessage(); | |
180 | |
181 // Updates |config_| and returns true if any change is made. | |
182 bool UpdateCurrentConfig(); | |
183 | |
170 rtc::scoped_ptr<FileWrapper> debug_file_; | 184 rtc::scoped_ptr<FileWrapper> debug_file_; |
171 rtc::scoped_ptr<audioproc::Event> event_msg_; // Protobuf message. | 185 rtc::scoped_ptr<audioproc::Event> event_msg_; // Protobuf message. |
172 std::string event_str_; // Memory for protobuf serialization. | 186 std::string event_str_; // Memory for protobuf serialization. |
187 rtc::scoped_ptr<audioproc::Config> config_; | |
173 #endif | 188 #endif |
174 | 189 |
175 // Format of processing streams at input/output call sites. | 190 // Format of processing streams at input/output call sites. |
176 ProcessingConfig api_format_; | 191 ProcessingConfig api_format_; |
177 | 192 |
178 // Only the rate and samples fields of fwd_proc_format_ are used because the | 193 // Only the rate and samples fields of fwd_proc_format_ are used because the |
179 // forward processing number of channels is mutable and is tracked by the | 194 // forward processing number of channels is mutable and is tracked by the |
180 // capture_audio_. | 195 // capture_audio_. |
181 StreamConfig fwd_proc_format_; | 196 StreamConfig fwd_proc_format_; |
182 StreamConfig rev_proc_format_; | 197 StreamConfig rev_proc_format_; |
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205 rtc::scoped_ptr<Beamformer<float>> beamformer_; | 220 rtc::scoped_ptr<Beamformer<float>> beamformer_; |
206 const std::vector<Point> array_geometry_; | 221 const std::vector<Point> array_geometry_; |
207 | 222 |
208 bool intelligibility_enabled_; | 223 bool intelligibility_enabled_; |
209 rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_; | 224 rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_; |
210 }; | 225 }; |
211 | 226 |
212 } // namespace webrtc | 227 } // namespace webrtc |
213 | 228 |
214 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ | 229 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ |
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