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Side by Side Diff: webrtc/audio_state.h

Issue 1403363003: Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly i… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: missing file Created 5 years, 1 month ago
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1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 *
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
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10 #ifndef WEBRTC_AUDIO_STATE_H_
11 #define WEBRTC_AUDIO_STATE_H_
12
13 #include "webrtc/base/linked_ptr.h"
14
15 namespace webrtc {
16
17 class AudioDeviceModule;
18 class VoiceEngine;
19
20 // AudioState holds the state which must be shared between multiple instances of
21 // webrtc::Call for audio processing purposes.
22 class AudioState {
23 public:
24 struct Config {
25 // VoiceEngine used for audio streams and audio/video synchronization.
26 // AudioState will tickle the VoE refcount to keep it alive for as long as
27 // the AudioState itself.
28 VoiceEngine* voice_engine = nullptr;
29
30 // The AudioDeviceModule associated with the Calls.
31 AudioDeviceModule* audio_device_module = nullptr;
32 };
33
34 static rtc::linked_ptr<AudioState> Create(const AudioState::Config& config);
35
36 virtual ~AudioState() {}
37 };
38 } // namespace webrtc
39
40 #endif // WEBRTC_AUDIO_STATE_H_
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