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https://github.com/mollyim/webrtc.git
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c89fdd716c
PlatformThread's API is using old style function pointers, causes casting, is unintuitive and forces artificial call sequences, and is additionally possible to misuse in release mode. Fix this by an API face lift: 1. The class is turned into a handle, which can be empty. 2. The only way of getting a non-empty PlatformThread is by calling SpawnJoinable or SpawnDetached, clearly conveying the semantics to the code reader. 3. Handles can be Finalized, which works differently for joinable and detached threads: a) Handles for detached threads are simply closed where applicable. b) Joinable threads are joined before handles are closed. 4. The destructor finalizes handles. No explicit call is needed. Fixed: webrtc:12727 Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33923}
208 lines
7.2 KiB
C++
208 lines
7.2 KiB
C++
/*
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* Copyright (c) 2012 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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#ifndef AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H_
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#define AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H_
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#include <memory>
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#include "modules/audio_device/audio_device_generic.h"
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#include "modules/audio_device/linux/audio_mixer_manager_alsa_linux.h"
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#include "rtc_base/platform_thread.h"
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#include "rtc_base/synchronization/mutex.h"
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#if defined(WEBRTC_USE_X11)
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#include <X11/Xlib.h>
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#endif
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#include <alsa/asoundlib.h>
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#include <sys/ioctl.h>
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#include <sys/soundcard.h>
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typedef webrtc::adm_linux_alsa::AlsaSymbolTable WebRTCAlsaSymbolTable;
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WebRTCAlsaSymbolTable* GetAlsaSymbolTable();
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namespace webrtc {
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class AudioDeviceLinuxALSA : public AudioDeviceGeneric {
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public:
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AudioDeviceLinuxALSA();
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virtual ~AudioDeviceLinuxALSA();
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// Retrieve the currently utilized audio layer
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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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InitStatus Init() RTC_LOCKS_EXCLUDED(mutex_) override;
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int32_t Terminate() RTC_LOCKS_EXCLUDED(mutex_) 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() RTC_LOCKS_EXCLUDED(mutex_) 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() RTC_LOCKS_EXCLUDED(mutex_) 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() RTC_LOCKS_EXCLUDED(mutex_) override;
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bool Playing() const override;
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int32_t StartRecording() override;
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int32_t StopRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
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bool Recording() const override;
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// Audio mixer initialization
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int32_t InitSpeaker() RTC_LOCKS_EXCLUDED(mutex_) override;
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bool SpeakerIsInitialized() const override;
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int32_t InitMicrophone() RTC_LOCKS_EXCLUDED(mutex_) 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)
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RTC_LOCKS_EXCLUDED(mutex_) 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)
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RTC_LOCKS_EXCLUDED(mutex_) 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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void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
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RTC_LOCKS_EXCLUDED(mutex_) override;
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private:
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int32_t InitRecordingLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
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int32_t StopRecordingLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
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int32_t StopPlayoutLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
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int32_t InitPlayoutLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
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int32_t InitSpeakerLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
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int32_t InitMicrophoneLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
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int32_t GetDevicesInfo(const int32_t function,
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const bool playback,
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const int32_t enumDeviceNo = 0,
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char* enumDeviceName = NULL,
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const int32_t ednLen = 0) const;
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int32_t ErrorRecovery(int32_t error, snd_pcm_t* deviceHandle);
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bool KeyPressed() const;
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void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(mutex_) { mutex_.Lock(); }
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void UnLock() RTC_UNLOCK_FUNCTION(mutex_) { mutex_.Unlock(); }
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inline int32_t InputSanityCheckAfterUnlockedPeriod() const;
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inline int32_t OutputSanityCheckAfterUnlockedPeriod() const;
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static void RecThreadFunc(void*);
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static void PlayThreadFunc(void*);
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bool RecThreadProcess();
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bool PlayThreadProcess();
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AudioDeviceBuffer* _ptrAudioBuffer;
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Mutex mutex_;
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rtc::PlatformThread _ptrThreadRec;
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rtc::PlatformThread _ptrThreadPlay;
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AudioMixerManagerLinuxALSA _mixerManager;
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uint16_t _inputDeviceIndex;
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uint16_t _outputDeviceIndex;
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bool _inputDeviceIsSpecified;
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bool _outputDeviceIsSpecified;
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snd_pcm_t* _handleRecord;
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snd_pcm_t* _handlePlayout;
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snd_pcm_uframes_t _recordingBuffersizeInFrame;
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snd_pcm_uframes_t _recordingPeriodSizeInFrame;
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snd_pcm_uframes_t _playoutBufferSizeInFrame;
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snd_pcm_uframes_t _playoutPeriodSizeInFrame;
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ssize_t _recordingBufferSizeIn10MS;
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ssize_t _playoutBufferSizeIn10MS;
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uint32_t _recordingFramesIn10MS;
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uint32_t _playoutFramesIn10MS;
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uint32_t _recordingFreq;
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uint32_t _playoutFreq;
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uint8_t _recChannels;
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uint8_t _playChannels;
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int8_t* _recordingBuffer; // in byte
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int8_t* _playoutBuffer; // in byte
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uint32_t _recordingFramesLeft;
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uint32_t _playoutFramesLeft;
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bool _initialized;
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bool _recording;
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bool _playing;
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bool _recIsInitialized;
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bool _playIsInitialized;
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snd_pcm_sframes_t _recordingDelay;
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snd_pcm_sframes_t _playoutDelay;
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char _oldKeyState[32];
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#if defined(WEBRTC_USE_X11)
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Display* _XDisplay;
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#endif
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_ALSA_LINUX_H_
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