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

Issue 2049683003: FileRecorder + FilePlayer: Let Create functions return unique_ptr (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@remove3
Patch Set: Created 4 years, 4 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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499 std::unique_ptr<RtpRtcp> _rtpRtcpModule; 499 std::unique_ptr<RtpRtcp> _rtpRtcpModule;
500 std::unique_ptr<AudioCodingModule> audio_coding_; 500 std::unique_ptr<AudioCodingModule> audio_coding_;
501 acm2::CodecManager codec_manager_; 501 acm2::CodecManager codec_manager_;
502 acm2::RentACodec rent_a_codec_; 502 acm2::RentACodec rent_a_codec_;
503 std::unique_ptr<AudioSinkInterface> audio_sink_; 503 std::unique_ptr<AudioSinkInterface> audio_sink_;
504 AudioLevel _outputAudioLevel; 504 AudioLevel _outputAudioLevel;
505 bool _externalTransport; 505 bool _externalTransport;
506 AudioFrame _audioFrame; 506 AudioFrame _audioFrame;
507 // Downsamples to the codec rate if necessary. 507 // Downsamples to the codec rate if necessary.
508 PushResampler<int16_t> input_resampler_; 508 PushResampler<int16_t> input_resampler_;
509 FilePlayer* _inputFilePlayerPtr; 509 std::unique_ptr<FilePlayer> input_file_player_;
510 FilePlayer* _outputFilePlayerPtr; 510 std::unique_ptr<FilePlayer> output_file_player_;
511 FileRecorder* _outputFileRecorderPtr; 511 std::unique_ptr<FileRecorder> output_file_recorder_;
512 int _inputFilePlayerId; 512 int _inputFilePlayerId;
513 int _outputFilePlayerId; 513 int _outputFilePlayerId;
514 int _outputFileRecorderId; 514 int _outputFileRecorderId;
515 bool _outputFileRecording; 515 bool _outputFileRecording;
516 bool _outputExternalMedia; 516 bool _outputExternalMedia;
517 VoEMediaProcess* _inputExternalMediaCallbackPtr; 517 VoEMediaProcess* _inputExternalMediaCallbackPtr;
518 VoEMediaProcess* _outputExternalMediaCallbackPtr; 518 VoEMediaProcess* _outputExternalMediaCallbackPtr;
519 uint32_t _timeStamp; 519 uint32_t _timeStamp;
520 520
521 RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_); 521 RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_);
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592 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; 592 std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
593 593
594 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. 594 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
595 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; 595 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
596 }; 596 };
597 597
598 } // namespace voe 598 } // namespace voe
599 } // namespace webrtc 599 } // namespace webrtc
600 600
601 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ 601 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_
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