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webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
Florent Castelli f4673f97ed Move webrtc::AudioDeviceModule include to api/ folder 
Bug: webrtc:15874
Change-Id: I5bdb19d5e710838b41e6ca283d406c9f1f21286b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348060
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42137}
2024-04-22 08:56:31 +00:00

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/*
* 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_PULSE_LINUX_H_
#define AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H_
#include <memory>
#include "api/audio/audio_device.h"
#include "api/audio/audio_device_defines.h"
#include "api/sequence_checker.h"
#include "modules/audio_device/audio_device_buffer.h"
#include "modules/audio_device/audio_device_generic.h"
#include "modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
#include "modules/audio_device/linux/pulseaudiosymboltable_linux.h"
#include "rtc_base/event.h"
#include "rtc_base/platform_thread.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
#if defined(WEBRTC_USE_X11)
#include <X11/Xlib.h>
#endif
#include <pulse/pulseaudio.h>
#include <stddef.h>
#include <stdint.h>
// We define this flag if it's missing from our headers, because we want to be
// able to compile against old headers but still use PA_STREAM_ADJUST_LATENCY
// if run against a recent version of the library.
#ifndef PA_STREAM_ADJUST_LATENCY
#define PA_STREAM_ADJUST_LATENCY 0x2000U
#endif
#ifndef PA_STREAM_START_MUTED
#define PA_STREAM_START_MUTED 0x1000U
#endif
// Set this constant to 0 to disable latency reading
const uint32_t WEBRTC_PA_REPORT_LATENCY = 1;
// Constants from implementation by Tristan Schmelcher [tschmelcher@google.com]
// First PulseAudio protocol version that supports PA_STREAM_ADJUST_LATENCY.
const uint32_t WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION = 13;
// Some timing constants for optimal operation. See
// https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008-January/001170.html
// for a good explanation of some of the factors that go into this.
// Playback.
// For playback, there is a round-trip delay to fill the server-side playback
// buffer, so setting too low of a latency is a buffer underflow risk. We will
// automatically increase the latency if a buffer underflow does occur, but we
// also enforce a sane minimum at start-up time. Anything lower would be
// virtually guaranteed to underflow at least once, so there's no point in
// allowing lower latencies.
const uint32_t WEBRTC_PA_PLAYBACK_LATENCY_MINIMUM_MSECS = 20;
// Every time a playback stream underflows, we will reconfigure it with target
// latency that is greater by this amount.
const uint32_t WEBRTC_PA_PLAYBACK_LATENCY_INCREMENT_MSECS = 20;
// We also need to configure a suitable request size. Too small and we'd burn
// CPU from the overhead of transfering small amounts of data at once. Too large
// and the amount of data remaining in the buffer right before refilling it
// would be a buffer underflow risk. We set it to half of the buffer size.
const uint32_t WEBRTC_PA_PLAYBACK_REQUEST_FACTOR = 2;
// Capture.
// For capture, low latency is not a buffer overflow risk, but it makes us burn
// CPU from the overhead of transfering small amounts of data at once, so we set
// a recommended value that we use for the kLowLatency constant (but if the user
// explicitly requests something lower then we will honour it).
// 1ms takes about 6-7% CPU. 5ms takes about 5%. 10ms takes about 4.x%.
const uint32_t WEBRTC_PA_LOW_CAPTURE_LATENCY_MSECS = 10;
// There is a round-trip delay to ack the data to the server, so the
// server-side buffer needs extra space to prevent buffer overflow. 20ms is
// sufficient, but there is no penalty to making it bigger, so we make it huge.
// (750ms is libpulse's default value for the _total_ buffer size in the
// kNoLatencyRequirements case.)
const uint32_t WEBRTC_PA_CAPTURE_BUFFER_EXTRA_MSECS = 750;
const uint32_t WEBRTC_PA_MSECS_PER_SEC = 1000;
// Init _configuredLatencyRec/Play to this value to disable latency requirements
const int32_t WEBRTC_PA_NO_LATENCY_REQUIREMENTS = -1;
// Set this const to 1 to account for peeked and used data in latency
// calculation
const uint32_t WEBRTC_PA_CAPTURE_BUFFER_LATENCY_ADJUSTMENT = 0;
typedef webrtc::adm_linux_pulse::PulseAudioSymbolTable WebRTCPulseSymbolTable;
WebRTCPulseSymbolTable* GetPulseSymbolTable();
namespace webrtc {
class AudioDeviceLinuxPulse : public AudioDeviceGeneric {
public:
AudioDeviceLinuxPulse();
virtual ~AudioDeviceLinuxPulse();
// Retrieve the currently utilized audio layer
int32_t ActiveAudioLayer(
AudioDeviceModule::AudioLayer& audioLayer) const override;
// Main initializaton and termination
InitStatus Init() 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() override;
bool RecordingIsInitialized() const override;
// Audio transport control
int32_t StartPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
int32_t StopPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
bool Playing() const override;
int32_t StartRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
int32_t StopRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
bool Recording() 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
RTC_LOCKS_EXCLUDED(mutex_) override;
void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
private:
void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(mutex_) { mutex_.Lock(); }
void UnLock() RTC_UNLOCK_FUNCTION(mutex_) { mutex_.Unlock(); }
void WaitForOperationCompletion(pa_operation* paOperation) const;
void WaitForSuccess(pa_operation* paOperation) const;
bool KeyPressed() const;
static void PaContextStateCallback(pa_context* c, void* pThis);
static void PaSinkInfoCallback(pa_context* c,
const pa_sink_info* i,
int eol,
void* pThis);
static void PaSourceInfoCallback(pa_context* c,
const pa_source_info* i,
int eol,
void* pThis);
static void PaServerInfoCallback(pa_context* c,
const pa_server_info* i,
void* pThis);
static void PaStreamStateCallback(pa_stream* p, void* pThis);
void PaContextStateCallbackHandler(pa_context* c);
void PaSinkInfoCallbackHandler(const pa_sink_info* i, int eol);
void PaSourceInfoCallbackHandler(const pa_source_info* i, int eol);
void PaServerInfoCallbackHandler(const pa_server_info* i);
void PaStreamStateCallbackHandler(pa_stream* p);
void EnableWriteCallback();
void DisableWriteCallback();
static void PaStreamWriteCallback(pa_stream* unused,
size_t buffer_space,
void* pThis);
void PaStreamWriteCallbackHandler(size_t buffer_space);
static void PaStreamUnderflowCallback(pa_stream* unused, void* pThis);
void PaStreamUnderflowCallbackHandler();
void EnableReadCallback();
void DisableReadCallback();
static void PaStreamReadCallback(pa_stream* unused1,
size_t unused2,
void* pThis);
void PaStreamReadCallbackHandler();
static void PaStreamOverflowCallback(pa_stream* unused, void* pThis);
void PaStreamOverflowCallbackHandler();
int32_t LatencyUsecs(pa_stream* stream);
int32_t ReadRecordedData(const void* bufferData, size_t bufferSize);
int32_t ProcessRecordedData(int8_t* bufferData,
uint32_t bufferSizeInSamples,
uint32_t recDelay);
int32_t CheckPulseAudioVersion();
int32_t InitSamplingFrequency();
int32_t GetDefaultDeviceInfo(bool recDevice, char* name, uint16_t& index);
int32_t InitPulseAudio();
int32_t TerminatePulseAudio();
void PaLock();
void PaUnLock();
static void RecThreadFunc(void*);
static void PlayThreadFunc(void*);
bool RecThreadProcess() RTC_LOCKS_EXCLUDED(mutex_);
bool PlayThreadProcess() RTC_LOCKS_EXCLUDED(mutex_);
AudioDeviceBuffer* _ptrAudioBuffer;
mutable Mutex mutex_;
rtc::Event _timeEventRec;
rtc::Event _timeEventPlay;
rtc::Event _recStartEvent;
rtc::Event _playStartEvent;
rtc::PlatformThread _ptrThreadPlay;
rtc::PlatformThread _ptrThreadRec;
AudioMixerManagerLinuxPulse _mixerManager;
uint16_t _inputDeviceIndex;
uint16_t _outputDeviceIndex;
bool _inputDeviceIsSpecified;
bool _outputDeviceIsSpecified;
int sample_rate_hz_;
uint8_t _recChannels;
uint8_t _playChannels;
// Stores thread ID in constructor.
// We can then use RTC_DCHECK_RUN_ON(&worker_thread_checker_) to ensure that
// other methods are called from the same thread.
// Currently only does RTC_DCHECK(thread_checker_.IsCurrent()).
SequenceChecker thread_checker_;
bool _initialized;
bool _recording;
bool _playing;
bool _recIsInitialized;
bool _playIsInitialized;
bool _startRec;
bool _startPlay;
bool update_speaker_volume_at_startup_;
bool quit_ RTC_GUARDED_BY(&mutex_);
uint32_t _sndCardPlayDelay RTC_GUARDED_BY(&mutex_);
int32_t _writeErrors;
uint16_t _deviceIndex;
int16_t _numPlayDevices;
int16_t _numRecDevices;
char* _playDeviceName;
char* _recDeviceName;
char* _playDisplayDeviceName;
char* _recDisplayDeviceName;
char _paServerVersion[32];
int8_t* _playBuffer;
size_t _playbackBufferSize;
size_t _playbackBufferUnused;
size_t _tempBufferSpace;
int8_t* _recBuffer;
size_t _recordBufferSize;
size_t _recordBufferUsed;
const void* _tempSampleData;
size_t _tempSampleDataSize;
int32_t _configuredLatencyPlay;
int32_t _configuredLatencyRec;
// PulseAudio
uint16_t _paDeviceIndex;
bool _paStateChanged;
pa_threaded_mainloop* _paMainloop;
pa_mainloop_api* _paMainloopApi;
pa_context* _paContext;
pa_stream* _recStream;
pa_stream* _playStream;
uint32_t _recStreamFlags;
uint32_t _playStreamFlags;
pa_buffer_attr _playBufferAttr;
pa_buffer_attr _recBufferAttr;
char _oldKeyState[32];
#if defined(WEBRTC_USE_X11)
Display* _XDisplay;
#endif
};
} // namespace webrtc
#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_PULSE_LINUX_H_
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