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Side by Side Diff: webrtc/voice_engine/channel.h

Issue 2622493002: Move FilePlayer and FileRecorder to Voice Engine (Closed)
Patch Set: Created 3 years, 11 months ago
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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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