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Side by Side Diff: webrtc/api/test/fakeaudiocapturemodule.h

Issue 1703833002: Remove ignored return code from modules. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 10 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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49 // Returns the number of frames that have been successfully pulled by the 49 // Returns the number of frames that have been successfully pulled by the
50 // instance. Note that correctly detecting success can only be done if the 50 // instance. Note that correctly detecting success can only be done if the
51 // pulled frame was generated/pushed from a FakeAudioCaptureModule. 51 // pulled frame was generated/pushed from a FakeAudioCaptureModule.
52 int frames_received() const; 52 int frames_received() const;
53 53
54 // Following functions are inherited from webrtc::AudioDeviceModule. 54 // Following functions are inherited from webrtc::AudioDeviceModule.
55 // Only functions called by PeerConnection are implemented, the rest do 55 // Only functions called by PeerConnection are implemented, the rest do
56 // nothing and return success. If a function is not expected to be called by 56 // nothing and return success. If a function is not expected to be called by
57 // PeerConnection an assertion is triggered if it is in fact called. 57 // PeerConnection an assertion is triggered if it is in fact called.
58 int64_t TimeUntilNextProcess() override; 58 int64_t TimeUntilNextProcess() override;
59 int32_t Process() override; 59 void Process() override;
60 60
61 int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override; 61 int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override;
62 62
63 ErrorCode LastError() const override; 63 ErrorCode LastError() const override;
64 int32_t RegisterEventObserver( 64 int32_t RegisterEventObserver(
65 webrtc::AudioDeviceObserver* event_callback) override; 65 webrtc::AudioDeviceObserver* event_callback) override;
66 66
67 // Note: Calling this method from a callback may result in deadlock. 67 // Note: Calling this method from a callback may result in deadlock.
68 int32_t RegisterAudioCallback( 68 int32_t RegisterAudioCallback(
69 webrtc::AudioTransport* audio_callback) override; 69 webrtc::AudioTransport* audio_callback) override;
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261 261
262 // Protects variables that are accessed from process_thread_ and 262 // Protects variables that are accessed from process_thread_ and
263 // the main thread. 263 // the main thread.
264 rtc::CriticalSection crit_; 264 rtc::CriticalSection crit_;
265 // Protects |audio_callback_| that is accessed from process_thread_ and 265 // Protects |audio_callback_| that is accessed from process_thread_ and
266 // the main thread. 266 // the main thread.
267 rtc::CriticalSection crit_callback_; 267 rtc::CriticalSection crit_callback_;
268 }; 268 };
269 269
270 #endif // WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_ 270 #endif // WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
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