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Side by Side Diff: pc/test/fakeaudiocapturemodule.h

Issue 3020493002: Remove AudioDeviceObserver and make ADM not inherit from the Module interface.
Patch Set: linux build error Created 3 years, 2 months ago
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1 /* 1 /*
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright 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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45 static const size_t kNumberBytesPerSample = sizeof(Sample); 45 static const size_t kNumberBytesPerSample = sizeof(Sample);
46 46
47 // Creates a FakeAudioCaptureModule or returns NULL on failure. 47 // Creates a FakeAudioCaptureModule or returns NULL on failure.
48 static rtc::scoped_refptr<FakeAudioCaptureModule> Create(); 48 static rtc::scoped_refptr<FakeAudioCaptureModule> Create();
49 49
50 // Returns the number of frames that have been successfully pulled by the 50 // Returns the number of frames that have been successfully pulled by the
51 // instance. Note that correctly detecting success can only be done if the 51 // instance. Note that correctly detecting success can only be done if the
52 // pulled frame was generated/pushed from a FakeAudioCaptureModule. 52 // pulled frame was generated/pushed from a FakeAudioCaptureModule.
53 int frames_received() const; 53 int frames_received() const;
54 54
55 // Following functions are inherited from webrtc::AudioDeviceModule.
56 // Only functions called by PeerConnection are implemented, the rest do
57 // nothing and return success. If a function is not expected to be called by
58 // PeerConnection an assertion is triggered if it is in fact called.
59 int64_t TimeUntilNextProcess() override;
60 void Process() override;
61
62 int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override; 55 int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override;
63 56
64 ErrorCode LastError() const override; 57 ErrorCode LastError() const override;
65 int32_t RegisterEventObserver(
66 webrtc::AudioDeviceObserver* event_callback) override;
67 58
68 // Note: Calling this method from a callback may result in deadlock. 59 // Note: Calling this method from a callback may result in deadlock.
69 int32_t RegisterAudioCallback( 60 int32_t RegisterAudioCallback(
70 webrtc::AudioTransport* audio_callback) override; 61 webrtc::AudioTransport* audio_callback) override;
71 62
72 int32_t Init() override; 63 int32_t Init() override;
73 int32_t Terminate() override; 64 int32_t Terminate() override;
74 bool Initialized() const override; 65 bool Initialized() const override;
75 66
76 int16_t PlayoutDevices() override; 67 int16_t PlayoutDevices() override;
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203 // Starts the periodic calling of ProcessFrame() in a thread safe way. 194 // Starts the periodic calling of ProcessFrame() in a thread safe way.
204 void StartProcessP(); 195 void StartProcessP();
205 // Periodcally called function that ensures that frames are pulled and pushed 196 // Periodcally called function that ensures that frames are pulled and pushed
206 // periodically if enabled/started. 197 // periodically if enabled/started.
207 void ProcessFrameP(); 198 void ProcessFrameP();
208 // Pulls frames from the registered webrtc::AudioTransport. 199 // Pulls frames from the registered webrtc::AudioTransport.
209 void ReceiveFrameP(); 200 void ReceiveFrameP();
210 // Pushes frames to the registered webrtc::AudioTransport. 201 // Pushes frames to the registered webrtc::AudioTransport.
211 void SendFrameP(); 202 void SendFrameP();
212 203
213 // The time in milliseconds when Process() was last called or 0 if no call
214 // has been made.
215 int64_t last_process_time_ms_;
216
217 // Callback for playout and recording. 204 // Callback for playout and recording.
218 webrtc::AudioTransport* audio_callback_; 205 webrtc::AudioTransport* audio_callback_;
219 206
220 bool recording_; // True when audio is being pushed from the instance. 207 bool recording_; // True when audio is being pushed from the instance.
221 bool playing_; // True when audio is being pulled by the instance. 208 bool playing_; // True when audio is being pulled by the instance.
222 209
223 bool play_is_initialized_; // True when the instance is ready to pull audio. 210 bool play_is_initialized_; // True when the instance is ready to pull audio.
224 bool rec_is_initialized_; // True when the instance is ready to push audio. 211 bool rec_is_initialized_; // True when the instance is ready to push audio.
225 212
226 // Input to and output from RecordedDataIsAvailable(..) makes it possible to 213 // Input to and output from RecordedDataIsAvailable(..) makes it possible to
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248 235
249 // Protects variables that are accessed from process_thread_ and 236 // Protects variables that are accessed from process_thread_ and
250 // the main thread. 237 // the main thread.
251 rtc::CriticalSection crit_; 238 rtc::CriticalSection crit_;
252 // Protects |audio_callback_| that is accessed from process_thread_ and 239 // Protects |audio_callback_| that is accessed from process_thread_ and
253 // the main thread. 240 // the main thread.
254 rtc::CriticalSection crit_callback_; 241 rtc::CriticalSection crit_callback_;
255 }; 242 };
256 243
257 #endif // PC_TEST_FAKEAUDIOCAPTUREMODULE_H_ 244 #endif // PC_TEST_FAKEAUDIOCAPTUREMODULE_H_
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