2014-06-11 14:12:04 +00:00
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/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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2017-09-15 06:47:31 +02:00
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#ifndef AUDIO_DEVICE_FILE_AUDIO_DEVICE_H_
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#define AUDIO_DEVICE_FILE_AUDIO_DEVICE_H_
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2014-06-11 14:12:04 +00:00
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#include <stdio.h>
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2016-02-24 05:00:36 -08:00
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#include <memory>
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2014-06-11 14:12:04 +00:00
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#include <string>
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2017-09-15 06:47:31 +02:00
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#include "modules/audio_device/audio_device_generic.h"
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#include "rtc_base/criticalsection.h"
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#include "rtc_base/timeutils.h"
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#include "system_wrappers/include/file_wrapper.h"
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2014-06-11 14:12:04 +00:00
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2015-11-26 17:45:47 +01:00
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namespace rtc {
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class PlatformThread;
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} // namespace rtc
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2014-06-11 14:12:04 +00:00
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namespace webrtc {
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class EventWrapper;
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// This is a fake audio device which plays audio from a file as its microphone
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// and plays out into a file.
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class FileAudioDevice : public AudioDeviceGeneric {
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public:
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// Constructs a file audio device with |id|. It will read audio from
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// |inputFilename| and record output audio to |outputFilename|.
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//
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// The input file should be a readable 48k stereo raw file, and the output
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// file should point to a writable location. The output format will also be
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// 48k stereo raw audio.
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2017-10-11 15:14:51 +02:00
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FileAudioDevice(const char* inputFilename,
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const char* outputFilename);
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2016-10-28 06:51:59 -07:00
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virtual ~FileAudioDevice();
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2014-06-11 14:12:04 +00:00
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// Retrieve the currently utilized audio layer
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2015-03-04 12:58:35 +00:00
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int32_t ActiveAudioLayer(
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AudioDeviceModule::AudioLayer& audioLayer) const override;
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// Main initializaton and termination
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2016-07-01 13:35:19 +02:00
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InitStatus Init() override;
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int32_t Terminate() override;
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bool Initialized() const override;
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// Device enumeration
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int16_t PlayoutDevices() override;
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int16_t RecordingDevices() override;
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int32_t PlayoutDeviceName(uint16_t index,
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]) override;
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int32_t RecordingDeviceName(uint16_t index,
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]) override;
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// Device selection
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int32_t SetPlayoutDevice(uint16_t index) override;
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int32_t SetPlayoutDevice(
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AudioDeviceModule::WindowsDeviceType device) override;
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int32_t SetRecordingDevice(uint16_t index) override;
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int32_t SetRecordingDevice(
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AudioDeviceModule::WindowsDeviceType device) override;
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// Audio transport initialization
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int32_t PlayoutIsAvailable(bool& available) override;
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int32_t InitPlayout() override;
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bool PlayoutIsInitialized() const override;
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int32_t RecordingIsAvailable(bool& available) override;
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int32_t InitRecording() override;
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bool RecordingIsInitialized() const override;
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// Audio transport control
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int32_t StartPlayout() override;
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int32_t StopPlayout() override;
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bool Playing() const override;
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int32_t StartRecording() override;
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int32_t StopRecording() override;
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bool Recording() const override;
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// Microphone Automatic Gain Control (AGC)
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int32_t SetAGC(bool enable) override;
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bool AGC() const override;
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// Audio mixer initialization
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int32_t InitSpeaker() override;
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bool SpeakerIsInitialized() const override;
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int32_t InitMicrophone() override;
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bool MicrophoneIsInitialized() const override;
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// Speaker volume controls
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int32_t SpeakerVolumeIsAvailable(bool& available) override;
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int32_t SetSpeakerVolume(uint32_t volume) override;
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int32_t SpeakerVolume(uint32_t& volume) const override;
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int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
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int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
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// Microphone volume controls
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int32_t MicrophoneVolumeIsAvailable(bool& available) override;
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int32_t SetMicrophoneVolume(uint32_t volume) override;
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int32_t MicrophoneVolume(uint32_t& volume) const override;
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int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
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int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
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// Speaker mute control
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int32_t SpeakerMuteIsAvailable(bool& available) override;
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int32_t SetSpeakerMute(bool enable) override;
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int32_t SpeakerMute(bool& enabled) const override;
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// Microphone mute control
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int32_t MicrophoneMuteIsAvailable(bool& available) override;
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int32_t SetMicrophoneMute(bool enable) override;
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int32_t MicrophoneMute(bool& enabled) const override;
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// Stereo support
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int32_t StereoPlayoutIsAvailable(bool& available) override;
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int32_t SetStereoPlayout(bool enable) override;
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int32_t StereoPlayout(bool& enabled) const override;
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int32_t StereoRecordingIsAvailable(bool& available) override;
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int32_t SetStereoRecording(bool enable) override;
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int32_t StereoRecording(bool& enabled) const override;
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// Delay information and control
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int32_t PlayoutDelay(uint16_t& delayMS) const override;
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2014-06-11 14:12:04 +00:00
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2015-03-04 12:58:35 +00:00
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void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
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private:
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static bool RecThreadFunc(void*);
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static bool PlayThreadFunc(void*);
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bool RecThreadProcess();
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bool PlayThreadProcess();
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int32_t _playout_index;
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int32_t _record_index;
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AudioDeviceBuffer* _ptrAudioBuffer;
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int8_t* _recordingBuffer; // In bytes.
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int8_t* _playoutBuffer; // In bytes.
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uint32_t _recordingFramesLeft;
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uint32_t _playoutFramesLeft;
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2017-03-30 01:14:41 -07:00
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rtc::CriticalSection _critSect;
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2014-06-11 14:12:04 +00:00
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Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
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size_t _recordingBufferSizeIn10MS;
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size_t _recordingFramesIn10MS;
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size_t _playoutFramesIn10MS;
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2014-06-11 14:12:04 +00:00
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2015-11-26 17:45:47 +01:00
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// TODO(pbos): Make plain members instead of pointers and stop resetting them.
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std::unique_ptr<rtc::PlatformThread> _ptrThreadRec;
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std::unique_ptr<rtc::PlatformThread> _ptrThreadPlay;
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2014-06-11 14:12:04 +00:00
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bool _playing;
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bool _recording;
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2016-11-15 05:37:58 -08:00
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int64_t _lastCallPlayoutMillis;
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int64_t _lastCallRecordMillis;
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2014-06-11 14:12:04 +00:00
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2016-10-28 06:51:59 -07:00
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FileWrapper& _outputFile;
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FileWrapper& _inputFile;
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2014-06-11 14:12:04 +00:00
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std::string _outputFilename;
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std::string _inputFilename;
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};
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} // namespace webrtc
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2017-09-15 06:47:31 +02:00
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#endif // AUDIO_DEVICE_FILE_AUDIO_DEVICE_H_
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