mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-13 05:40:42 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
Markus Handell ad5037b4a8 Reland "Refactor the PlatformThread API." 
This reverts commit 793bac569f.

Reason for revert: rare compilation error fixed

Original change's description:
> Revert "Refactor the PlatformThread API."
>
> This reverts commit c89fdd716c.
>
> Reason for revert: Causes rare compilation error on win-libfuzzer-asan trybot.
> See https://ci.chromium.org/p/chromium/builders/try/win-libfuzzer-asan-rel/713745?
>
> Original change's description:
> > Refactor the PlatformThread API.
> >
> > 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}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=handellm@webrtc.org
>
> Bug: webrtc:12727
> Change-Id: Ic0146be8866f6dd3ad9c364fb8646650b8e07419
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217583
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33936}

# Not skipping CQ checks because this is a reland.

Bug: webrtc:12727
Change-Id: Ifd6f44eac72fed84474277a1be03eb84d2f4376e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217881
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33950}
2021-05-07 14:14:43 +00:00

208 lines
7.2 KiB
C++
Raw Blame History

/*
* Copyright (c) 2012 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_AUDIO_DEVICE_ALSA_LINUX_H_
#define AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H_
#include <memory>
#include "modules/audio_device/audio_device_generic.h"
#include "modules/audio_device/linux/audio_mixer_manager_alsa_linux.h"
#include "rtc_base/platform_thread.h"
#include "rtc_base/synchronization/mutex.h"
#if defined(WEBRTC_USE_X11)
#include <X11/Xlib.h>
#endif
#include <alsa/asoundlib.h>
#include <sys/ioctl.h>
#include <sys/soundcard.h>
typedef webrtc::adm_linux_alsa::AlsaSymbolTable WebRTCAlsaSymbolTable;
WebRTCAlsaSymbolTable* GetAlsaSymbolTable();
namespace webrtc {
class AudioDeviceLinuxALSA : public AudioDeviceGeneric {
public:
AudioDeviceLinuxALSA();
virtual ~AudioDeviceLinuxALSA();
// Retrieve the currently utilized audio layer
int32_t ActiveAudioLayer(
AudioDeviceModule::AudioLayer& audioLayer) const override;
// Main initializaton and termination
InitStatus Init() RTC_LOCKS_EXCLUDED(mutex_) override;
int32_t Terminate() RTC_LOCKS_EXCLUDED(mutex_) 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() RTC_LOCKS_EXCLUDED(mutex_) override;
bool PlayoutIsInitialized() const override;
int32_t RecordingIsAvailable(bool& available) override;
int32_t InitRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
bool RecordingIsInitialized() const override;
// Audio transport control
int32_t StartPlayout() override;
int32_t StopPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
bool Playing() const override;
int32_t StartRecording() override;
int32_t StopRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
bool Recording() const override;
// Audio mixer initialization
int32_t InitSpeaker() RTC_LOCKS_EXCLUDED(mutex_) override;
bool SpeakerIsInitialized() const override;
int32_t InitMicrophone() RTC_LOCKS_EXCLUDED(mutex_) 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)
RTC_LOCKS_EXCLUDED(mutex_) override;
int32_t SetStereoPlayout(bool enable) override;
int32_t StereoPlayout(bool& enabled) const override;
int32_t StereoRecordingIsAvailable(bool& available)
RTC_LOCKS_EXCLUDED(mutex_) 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)
RTC_LOCKS_EXCLUDED(mutex_) override;
private:
int32_t InitRecordingLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
int32_t StopRecordingLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
int32_t StopPlayoutLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
int32_t InitPlayoutLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
int32_t InitSpeakerLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
int32_t InitMicrophoneLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
int32_t GetDevicesInfo(const int32_t function,
const bool playback,
const int32_t enumDeviceNo = 0,
char* enumDeviceName = NULL,
const int32_t ednLen = 0) const;
int32_t ErrorRecovery(int32_t error, snd_pcm_t* deviceHandle);
bool KeyPressed() const;
void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(mutex_) { mutex_.Lock(); }
void UnLock() RTC_UNLOCK_FUNCTION(mutex_) { mutex_.Unlock(); }
inline int32_t InputSanityCheckAfterUnlockedPeriod() const;
inline int32_t OutputSanityCheckAfterUnlockedPeriod() const;
static void RecThreadFunc(void*);
static void PlayThreadFunc(void*);
bool RecThreadProcess();
bool PlayThreadProcess();
AudioDeviceBuffer* _ptrAudioBuffer;
Mutex mutex_;
rtc::PlatformThread _ptrThreadRec;
rtc::PlatformThread _ptrThreadPlay;
AudioMixerManagerLinuxALSA _mixerManager;
uint16_t _inputDeviceIndex;
uint16_t _outputDeviceIndex;
bool _inputDeviceIsSpecified;
bool _outputDeviceIsSpecified;
snd_pcm_t* _handleRecord;
snd_pcm_t* _handlePlayout;
snd_pcm_uframes_t _recordingBuffersizeInFrame;
snd_pcm_uframes_t _recordingPeriodSizeInFrame;
snd_pcm_uframes_t _playoutBufferSizeInFrame;
snd_pcm_uframes_t _playoutPeriodSizeInFrame;
ssize_t _recordingBufferSizeIn10MS;
ssize_t _playoutBufferSizeIn10MS;
uint32_t _recordingFramesIn10MS;
uint32_t _playoutFramesIn10MS;
uint32_t _recordingFreq;
uint32_t _playoutFreq;
uint8_t _recChannels;
uint8_t _playChannels;
int8_t* _recordingBuffer; // in byte
int8_t* _playoutBuffer; // in byte
uint32_t _recordingFramesLeft;
uint32_t _playoutFramesLeft;
bool _initialized;
bool _recording;
bool _playing;
bool _recIsInitialized;
bool _playIsInitialized;
snd_pcm_sframes_t _recordingDelay;
snd_pcm_sframes_t _playoutDelay;
char _oldKeyState[32];
#if defined(WEBRTC_USE_X11)
Display* _XDisplay;
#endif
};
} // namespace webrtc
#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_ALSA_LINUX_H_
Reference in a new issue View git blame Copy permalink