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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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25 #include "webrtc/voice_engine/monitor_module.h" | 25 #include "webrtc/voice_engine/monitor_module.h" |
26 #include "webrtc/voice_engine/voice_engine_defines.h" | 26 #include "webrtc/voice_engine/voice_engine_defines.h" |
27 | 27 |
28 #if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS) | 28 #if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS) |
29 #define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 1 | 29 #define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 1 |
30 #else | 30 #else |
31 #define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 0 | 31 #define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 0 |
32 #endif | 32 #endif |
33 | 33 |
34 namespace webrtc { | 34 namespace webrtc { |
35 | |
36 class AudioProcessing; | 35 class AudioProcessing; |
37 class ProcessThread; | 36 class ProcessThread; |
38 class VoEExternalMedia; | 37 class VoEExternalMedia; |
39 class VoEMediaProcess; | 38 class VoEMediaProcess; |
40 | 39 |
41 namespace voe { | 40 namespace voe { |
42 | 41 |
43 class ChannelManager; | 42 class ChannelManager; |
44 class MixedAudio; | 43 class MixedAudio; |
45 class Statistics; | 44 class Statistics; |
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60 | 59 |
61 int32_t PrepareDemux(const void* audioSamples, | 60 int32_t PrepareDemux(const void* audioSamples, |
62 size_t nSamples, | 61 size_t nSamples, |
63 size_t nChannels, | 62 size_t nChannels, |
64 uint32_t samplesPerSec, | 63 uint32_t samplesPerSec, |
65 uint16_t totalDelayMS, | 64 uint16_t totalDelayMS, |
66 int32_t clockDrift, | 65 int32_t clockDrift, |
67 uint16_t currentMicLevel, | 66 uint16_t currentMicLevel, |
68 bool keyPressed); | 67 bool keyPressed); |
69 | 68 |
70 | 69 void ProcessAudio(); |
71 int32_t DemuxAndMix(); | |
72 // Used by the Chrome to pass the recording data to the specific VoE | 70 // Used by the Chrome to pass the recording data to the specific VoE |
73 // channels for demux. | 71 // channels for processing. |
74 void DemuxAndMix(const int voe_channels[], size_t number_of_voe_channels); | 72 void ProcessAudio(const int voe_channels[], size_t number_of_voe_channels); |
75 | |
76 int32_t EncodeAndSend(); | |
77 // Used by the Chrome to pass the recording data to the specific VoE | |
78 // channels for encoding and sending to the network. | |
79 void EncodeAndSend(const int voe_channels[], size_t number_of_voe_channels); | |
80 | 73 |
81 // Must be called on the same thread as PrepareDemux(). | 74 // Must be called on the same thread as PrepareDemux(). |
82 uint32_t CaptureLevel() const; | 75 uint32_t CaptureLevel() const; |
83 | 76 |
84 int32_t StopSend(); | 77 int32_t StopSend(); |
85 | 78 |
86 // VoEExternalMedia | 79 // VoEExternalMedia |
87 int RegisterExternalMediaProcessing(VoEMediaProcess* object, | 80 int RegisterExternalMediaProcessing(VoEMediaProcess* object, |
88 ProcessingTypes type); | 81 ProcessingTypes type); |
89 int DeRegisterExternalMediaProcessing(ProcessingTypes type); | 82 int DeRegisterExternalMediaProcessing(ProcessingTypes type); |
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233 bool _mute; | 226 bool _mute; |
234 bool stereo_codec_; | 227 bool stereo_codec_; |
235 bool swap_stereo_channels_; | 228 bool swap_stereo_channels_; |
236 }; | 229 }; |
237 | 230 |
238 } // namespace voe | 231 } // namespace voe |
239 | 232 |
240 } // namespace webrtc | 233 } // namespace webrtc |
241 | 234 |
242 #endif // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H | 235 #endif // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H |
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