webrtc_m130/modules/audio_device/dummy/file_audio_device.h

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/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef AUDIO_DEVICE_FILE_AUDIO_DEVICE_H_
#define AUDIO_DEVICE_FILE_AUDIO_DEVICE_H_
#include <stdio.h>
#include <memory>
#include <string>
#include "modules/audio_device/audio_device_generic.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/file_wrapper.h"
namespace rtc {
class PlatformThread;
} // namespace rtc
namespace webrtc {
class EventWrapper;
// This is a fake audio device which plays audio from a file as its microphone
// and plays out into a file.
class FileAudioDevice : public AudioDeviceGeneric {
public:
// Constructs a file audio device with |id|. It will read audio from
// |inputFilename| and record output audio to |outputFilename|.
//
// The input file should be a readable 48k stereo raw file, and the output
// file should point to a writable location. The output format will also be
// 48k stereo raw audio.
FileAudioDevice(const char* inputFilename,
const char* outputFilename);
virtual ~FileAudioDevice();
// Retrieve the currently utilized audio layer
int32_t ActiveAudioLayer(
AudioDeviceModule::AudioLayer& audioLayer) const override;
// Main initializaton and termination
InitStatus Init() override;
int32_t Terminate() override;
bool Initialized() const override;
// Device enumeration
int16_t PlayoutDevices() override;
int16_t RecordingDevices() override;
int32_t PlayoutDeviceName(uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) override;
int32_t RecordingDeviceName(uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) override;
// Device selection
int32_t SetPlayoutDevice(uint16_t index) override;
int32_t SetPlayoutDevice(
AudioDeviceModule::WindowsDeviceType device) override;
int32_t SetRecordingDevice(uint16_t index) override;
int32_t SetRecordingDevice(
AudioDeviceModule::WindowsDeviceType device) override;
// Audio transport initialization
int32_t PlayoutIsAvailable(bool& available) override;
int32_t InitPlayout() override;
bool PlayoutIsInitialized() const override;
int32_t RecordingIsAvailable(bool& available) override;
int32_t InitRecording() override;
bool RecordingIsInitialized() const override;
// Audio transport control
int32_t StartPlayout() override;
int32_t StopPlayout() override;
bool Playing() const override;
int32_t StartRecording() override;
int32_t StopRecording() override;
bool Recording() const override;
// Microphone Automatic Gain Control (AGC)
int32_t SetAGC(bool enable) override;
bool AGC() const override;
// Audio mixer initialization
int32_t InitSpeaker() override;
bool SpeakerIsInitialized() const override;
int32_t InitMicrophone() override;
bool MicrophoneIsInitialized() const override;
// Speaker volume controls
int32_t SpeakerVolumeIsAvailable(bool& available) override;
int32_t SetSpeakerVolume(uint32_t volume) override;
int32_t SpeakerVolume(uint32_t& volume) const override;
int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
// Microphone volume controls
int32_t MicrophoneVolumeIsAvailable(bool& available) override;
int32_t SetMicrophoneVolume(uint32_t volume) override;
int32_t MicrophoneVolume(uint32_t& volume) const override;
int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
// Speaker mute control
int32_t SpeakerMuteIsAvailable(bool& available) override;
int32_t SetSpeakerMute(bool enable) override;
int32_t SpeakerMute(bool& enabled) const override;
// Microphone mute control
int32_t MicrophoneMuteIsAvailable(bool& available) override;
int32_t SetMicrophoneMute(bool enable) override;
int32_t MicrophoneMute(bool& enabled) const override;
// Stereo support
int32_t StereoPlayoutIsAvailable(bool& available) override;
int32_t SetStereoPlayout(bool enable) override;
int32_t StereoPlayout(bool& enabled) const override;
int32_t StereoRecordingIsAvailable(bool& available) override;
int32_t SetStereoRecording(bool enable) override;
int32_t StereoRecording(bool& enabled) const override;
// Delay information and control
int32_t PlayoutDelay(uint16_t& delayMS) const override;
void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
private:
static bool RecThreadFunc(void*);
static bool PlayThreadFunc(void*);
bool RecThreadProcess();
bool PlayThreadProcess();
int32_t _playout_index;
int32_t _record_index;
AudioDeviceBuffer* _ptrAudioBuffer;
int8_t* _recordingBuffer; // In bytes.
int8_t* _playoutBuffer; // In bytes.
uint32_t _recordingFramesLeft;
uint32_t _playoutFramesLeft;
rtc::CriticalSection _critSect;
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
size_t _recordingBufferSizeIn10MS;
size_t _recordingFramesIn10MS;
size_t _playoutFramesIn10MS;
// TODO(pbos): Make plain members instead of pointers and stop resetting them.
std::unique_ptr<rtc::PlatformThread> _ptrThreadRec;
std::unique_ptr<rtc::PlatformThread> _ptrThreadPlay;
bool _playing;
bool _recording;
int64_t _lastCallPlayoutMillis;
int64_t _lastCallRecordMillis;
FileWrapper& _outputFile;
FileWrapper& _inputFile;
std::string _outputFilename;
std::string _inputFilename;
};
} // namespace webrtc
#endif // AUDIO_DEVICE_FILE_AUDIO_DEVICE_H_