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

Issue 2059703002: Move webrtc/audio_*.h to webrtc/api/call (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased on top of d565b73121b1b7672fb7d1f115bbbbb137a838eb Created 4 years, 3 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
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/audio_sink.h" 16 #include "webrtc/api/call/audio_sink.h"
17 #include "webrtc/base/criticalsection.h" 17 #include "webrtc/base/criticalsection.h"
18 #include "webrtc/base/optional.h" 18 #include "webrtc/base/optional.h"
19 #include "webrtc/common_audio/resampler/include/push_resampler.h" 19 #include "webrtc/common_audio/resampler/include/push_resampler.h"
20 #include "webrtc/common_types.h" 20 #include "webrtc/common_types.h"
21 #include "webrtc/modules/audio_coding/acm2/codec_manager.h" 21 #include "webrtc/modules/audio_coding/acm2/codec_manager.h"
22 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" 22 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
23 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" 23 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
24 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d efines.h" 24 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d efines.h"
25 #include "webrtc/modules/audio_processing/rms_level.h" 25 #include "webrtc/modules/audio_processing/rms_level.h"
26 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" 26 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
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588 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; 588 std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
589 589
590 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. 590 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
591 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; 591 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
592 }; 592 };
593 593
594 } // namespace voe 594 } // namespace voe
595 } // namespace webrtc 595 } // namespace webrtc
596 596
597 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ 597 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_
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