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Side by Side Diff: webrtc/modules/audio_processing/audio_processing_impl.h

Issue 1348903004: Adding APM configuration in AEC dump. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: last commit was hasty, having an error Created 5 years, 2 months ago
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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
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160 std::list<ProcessingComponent*> component_list_; 160 std::list<ProcessingComponent*> component_list_;
161 CriticalSectionWrapper* crit_; 161 CriticalSectionWrapper* crit_;
162 rtc::scoped_ptr<AudioBuffer> render_audio_; 162 rtc::scoped_ptr<AudioBuffer> render_audio_;
163 rtc::scoped_ptr<AudioBuffer> capture_audio_; 163 rtc::scoped_ptr<AudioBuffer> capture_audio_;
164 rtc::scoped_ptr<AudioConverter> render_converter_; 164 rtc::scoped_ptr<AudioConverter> render_converter_;
165 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP 165 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
166 // TODO(andrew): make this more graceful. Ideally we would split this stuff 166 // TODO(andrew): make this more graceful. Ideally we would split this stuff
167 // out into a separate class with an "enabled" and "disabled" implementation. 167 // out into a separate class with an "enabled" and "disabled" implementation.
168 int WriteMessageToDebugFile(); 168 int WriteMessageToDebugFile();
169 int WriteInitMessage(); 169 int WriteInitMessage();
170
171 // Writes Config message. If not |forced|, only writes the current config if
172 // it is different from the last saved one; if |forced|, writes the config
173 // regardless of the last saved.
174 int WriteConfigMessage(bool forced);
175
170 rtc::scoped_ptr<FileWrapper> debug_file_; 176 rtc::scoped_ptr<FileWrapper> debug_file_;
171 rtc::scoped_ptr<audioproc::Event> event_msg_; // Protobuf message. 177 rtc::scoped_ptr<audioproc::Event> event_msg_; // Protobuf message.
172 std::string event_str_; // Memory for protobuf serialization. 178 std::string event_str_; // Memory for protobuf serialization.
179
180 // Serialized string of last saved APM configuration.
181 std::string last_serialized_config_;
173 #endif 182 #endif
174 183
175 // Format of processing streams at input/output call sites. 184 // Format of processing streams at input/output call sites.
176 ProcessingConfig api_format_; 185 ProcessingConfig api_format_;
177 186
178 // Only the rate and samples fields of fwd_proc_format_ are used because the 187 // 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 188 // forward processing number of channels is mutable and is tracked by the
180 // capture_audio_. 189 // capture_audio_.
181 StreamConfig fwd_proc_format_; 190 StreamConfig fwd_proc_format_;
182 StreamConfig rev_proc_format_; 191 StreamConfig rev_proc_format_;
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205 rtc::scoped_ptr<Beamformer<float>> beamformer_; 214 rtc::scoped_ptr<Beamformer<float>> beamformer_;
206 const std::vector<Point> array_geometry_; 215 const std::vector<Point> array_geometry_;
207 216
208 bool intelligibility_enabled_; 217 bool intelligibility_enabled_;
209 rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_; 218 rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_;
210 }; 219 };
211 220
212 } // namespace webrtc 221 } // namespace webrtc
213 222
214 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ 223 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
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