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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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21 #include "webrtc/common_audio/resampler/include/push_resampler.h" | 21 #include "webrtc/common_audio/resampler/include/push_resampler.h" |
22 #include "webrtc/common_types.h" | 22 #include "webrtc/common_types.h" |
23 #include "webrtc/modules/audio_coding/acm2/codec_manager.h" | 23 #include "webrtc/modules/audio_coding/acm2/codec_manager.h" |
24 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" | 24 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" |
25 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" | 25 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
26 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d
efines.h" | 26 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d
efines.h" |
27 #include "webrtc/modules/audio_processing/rms_level.h" | 27 #include "webrtc/modules/audio_processing/rms_level.h" |
28 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" | 28 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
29 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 29 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
30 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 30 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
31 #include "webrtc/modules/utility/include/file_player.h" | 31 #include "webrtc/voice_engine/file_player.h" |
32 #include "webrtc/modules/utility/include/file_recorder.h" | 32 #include "webrtc/voice_engine/file_recorder.h" |
33 #include "webrtc/voice_engine/include/voe_audio_processing.h" | 33 #include "webrtc/voice_engine/include/voe_audio_processing.h" |
34 #include "webrtc/voice_engine/include/voe_base.h" | 34 #include "webrtc/voice_engine/include/voe_base.h" |
35 #include "webrtc/voice_engine/include/voe_network.h" | 35 #include "webrtc/voice_engine/include/voe_network.h" |
36 #include "webrtc/voice_engine/level_indicator.h" | 36 #include "webrtc/voice_engine/level_indicator.h" |
37 #include "webrtc/voice_engine/shared_data.h" | 37 #include "webrtc/voice_engine/shared_data.h" |
38 #include "webrtc/voice_engine/voice_engine_defines.h" | 38 #include "webrtc/voice_engine/voice_engine_defines.h" |
39 | 39 |
40 namespace rtc { | 40 namespace rtc { |
41 class TimestampWrapAroundHandler; | 41 class TimestampWrapAroundHandler; |
42 } | 42 } |
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551 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. | 551 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. |
552 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 552 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
553 | 553 |
554 SmoothingFilterImpl bitrate_smoother_; | 554 SmoothingFilterImpl bitrate_smoother_; |
555 }; | 555 }; |
556 | 556 |
557 } // namespace voe | 557 } // namespace voe |
558 } // namespace webrtc | 558 } // namespace webrtc |
559 | 559 |
560 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 560 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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