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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 |
11 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 11 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
12 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 12 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 | 15 |
16 #include "webrtc/api/audio/audio_mixer.h" | 16 #include "webrtc/api/audio/audio_mixer.h" |
17 #include "webrtc/api/audio_codecs/audio_encoder.h" | 17 #include "webrtc/api/audio_codecs/audio_encoder.h" |
18 #include "webrtc/api/call/audio_sink.h" | 18 #include "webrtc/api/call/audio_sink.h" |
| 19 #include "webrtc/base/criticalsection.h" |
| 20 #include "webrtc/base/event.h" |
| 21 #include "webrtc/base/optional.h" |
| 22 #include "webrtc/base/thread_checker.h" |
19 #include "webrtc/common_audio/resampler/include/push_resampler.h" | 23 #include "webrtc/common_audio/resampler/include/push_resampler.h" |
20 #include "webrtc/common_types.h" | 24 #include "webrtc/common_types.h" |
21 #include "webrtc/modules/audio_coding/acm2/codec_manager.h" | 25 #include "webrtc/modules/audio_coding/acm2/codec_manager.h" |
22 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" | 26 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" |
23 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" | 27 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
24 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d
efines.h" | 28 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d
efines.h" |
25 #include "webrtc/modules/audio_processing/rms_level.h" | 29 #include "webrtc/modules/audio_processing/rms_level.h" |
26 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" | 30 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
27 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 31 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
28 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" | 32 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" |
29 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 33 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
30 #include "webrtc/rtc_base/criticalsection.h" | |
31 #include "webrtc/rtc_base/event.h" | |
32 #include "webrtc/rtc_base/optional.h" | |
33 #include "webrtc/rtc_base/thread_checker.h" | |
34 #include "webrtc/voice_engine/audio_level.h" | 34 #include "webrtc/voice_engine/audio_level.h" |
35 #include "webrtc/voice_engine/file_player.h" | 35 #include "webrtc/voice_engine/file_player.h" |
36 #include "webrtc/voice_engine/file_recorder.h" | 36 #include "webrtc/voice_engine/file_recorder.h" |
37 #include "webrtc/voice_engine/include/voe_base.h" | 37 #include "webrtc/voice_engine/include/voe_base.h" |
38 #include "webrtc/voice_engine/include/voe_network.h" | 38 #include "webrtc/voice_engine/include/voe_network.h" |
39 #include "webrtc/voice_engine/shared_data.h" | 39 #include "webrtc/voice_engine/shared_data.h" |
40 #include "webrtc/voice_engine/voice_engine_defines.h" | 40 #include "webrtc/voice_engine/voice_engine_defines.h" |
41 | 41 |
42 namespace rtc { | 42 namespace rtc { |
43 class TimestampWrapAroundHandler; | 43 class TimestampWrapAroundHandler; |
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553 | 553 |
554 bool encoder_queue_is_active_ GUARDED_BY(encoder_queue_lock_) = false; | 554 bool encoder_queue_is_active_ GUARDED_BY(encoder_queue_lock_) = false; |
555 | 555 |
556 rtc::TaskQueue* encoder_queue_ = nullptr; | 556 rtc::TaskQueue* encoder_queue_ = nullptr; |
557 }; | 557 }; |
558 | 558 |
559 } // namespace voe | 559 } // namespace voe |
560 } // namespace webrtc | 560 } // namespace webrtc |
561 | 561 |
562 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 562 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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