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

Issue 2037623002: Move FilePlayer and FileRecorder to Voice Engine (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@remove0
Patch Set: GN build fix Created 4 years, 6 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/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"
27 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" 27 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
28 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" 28 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
29 #include "webrtc/modules/utility/include/file_player.h" 29 #include "webrtc/voice_engine/file_player.h"
30 #include "webrtc/modules/utility/include/file_recorder.h" 30 #include "webrtc/voice_engine/file_recorder.h"
31 #include "webrtc/voice_engine/include/voe_audio_processing.h" 31 #include "webrtc/voice_engine/include/voe_audio_processing.h"
32 #include "webrtc/voice_engine/include/voe_network.h" 32 #include "webrtc/voice_engine/include/voe_network.h"
33 #include "webrtc/voice_engine/level_indicator.h" 33 #include "webrtc/voice_engine/level_indicator.h"
34 #include "webrtc/voice_engine/network_predictor.h" 34 #include "webrtc/voice_engine/network_predictor.h"
35 #include "webrtc/voice_engine/shared_data.h" 35 #include "webrtc/voice_engine/shared_data.h"
36 #include "webrtc/voice_engine/voice_engine_defines.h" 36 #include "webrtc/voice_engine/voice_engine_defines.h"
37 37
38 namespace rtc { 38 namespace rtc {
39 39
40 class TimestampWrapAroundHandler; 40 class TimestampWrapAroundHandler;
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589 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; 589 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
590 590
591 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. 591 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
592 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; 592 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
593 }; 593 };
594 594
595 } // namespace voe 595 } // namespace voe
596 } // namespace webrtc 596 } // namespace webrtc
597 597
598 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ 598 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_
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