blob: cde7e4643902432579babb6269d349d81eb3b106 [file] [log] [blame]
* 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.
#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include <X11/Xlib.h>
#include <pulse/pulseaudio.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.
// Set this constant to 0 to disable latency reading
const uint32_t WEBRTC_PA_REPORT_LATENCY = 1;
// Constants from implementation by Tristan Schmelcher []
// First PulseAudio protocol version that supports PA_STREAM_ADJUST_LATENCY.
// Some timing constants for optimal operation. See
// 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.
// Every time a playback stream underflows, we will reconfigure it with target
// latency that is greater by this amount.
// 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.
// 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%.
// 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_MSECS_PER_SEC = 1000;
// Init _configuredLatencyRec/Play to this value to disable latency requirements
// Set this const to 1 to account for peeked and used data in latency calculation
namespace webrtc
class EventWrapper;
class ThreadWrapper;
class AudioDeviceLinuxPulse: public AudioDeviceGeneric
AudioDeviceLinuxPulse(const int32_t id);
virtual ~AudioDeviceLinuxPulse();
// Retrieve the currently utilized audio layer
virtual int32_t ActiveAudioLayer(
AudioDeviceModule::AudioLayer& audioLayer) const OVERRIDE;
// Main initializaton and termination
virtual int32_t Init() OVERRIDE;
virtual int32_t Terminate() OVERRIDE;
virtual bool Initialized() const OVERRIDE;
// Device enumeration
virtual int16_t PlayoutDevices() OVERRIDE;
virtual int16_t RecordingDevices() OVERRIDE;
virtual int32_t PlayoutDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) OVERRIDE;
virtual int32_t RecordingDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) OVERRIDE;
// Device selection
virtual int32_t SetPlayoutDevice(uint16_t index) OVERRIDE;
virtual int32_t SetPlayoutDevice(
AudioDeviceModule::WindowsDeviceType device) OVERRIDE;
virtual int32_t SetRecordingDevice(uint16_t index) OVERRIDE;
virtual int32_t SetRecordingDevice(
AudioDeviceModule::WindowsDeviceType device) OVERRIDE;
// Audio transport initialization
virtual int32_t PlayoutIsAvailable(bool& available) OVERRIDE;
virtual int32_t InitPlayout() OVERRIDE;
virtual bool PlayoutIsInitialized() const OVERRIDE;
virtual int32_t RecordingIsAvailable(bool& available) OVERRIDE;
virtual int32_t InitRecording() OVERRIDE;
virtual bool RecordingIsInitialized() const OVERRIDE;
// Audio transport control
virtual int32_t StartPlayout() OVERRIDE;
virtual int32_t StopPlayout() OVERRIDE;
virtual bool Playing() const OVERRIDE;
virtual int32_t StartRecording() OVERRIDE;
virtual int32_t StopRecording() OVERRIDE;
virtual bool Recording() const OVERRIDE;
// Microphone Automatic Gain Control (AGC)
virtual int32_t SetAGC(bool enable) OVERRIDE;
virtual bool AGC() const OVERRIDE;
// Volume control based on the Windows Wave API (Windows only)
virtual int32_t SetWaveOutVolume(uint16_t volumeLeft,
uint16_t volumeRight) OVERRIDE;
virtual int32_t WaveOutVolume(uint16_t& volumeLeft,
uint16_t& volumeRight) const OVERRIDE;
// Audio mixer initialization
virtual int32_t InitSpeaker() OVERRIDE;
virtual bool SpeakerIsInitialized() const OVERRIDE;
virtual int32_t InitMicrophone() OVERRIDE;
virtual bool MicrophoneIsInitialized() const OVERRIDE;
// Speaker volume controls
virtual int32_t SpeakerVolumeIsAvailable(bool& available) OVERRIDE;
virtual int32_t SetSpeakerVolume(uint32_t volume) OVERRIDE;
virtual int32_t SpeakerVolume(uint32_t& volume) const OVERRIDE;
virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const OVERRIDE;
virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const OVERRIDE;
virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const OVERRIDE;
// Microphone volume controls
virtual int32_t MicrophoneVolumeIsAvailable(bool& available) OVERRIDE;
virtual int32_t SetMicrophoneVolume(uint32_t volume) OVERRIDE;
virtual int32_t MicrophoneVolume(uint32_t& volume) const OVERRIDE;
virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const OVERRIDE;
virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const OVERRIDE;
virtual int32_t MicrophoneVolumeStepSize(
uint16_t& stepSize) const OVERRIDE;
// Speaker mute control
virtual int32_t SpeakerMuteIsAvailable(bool& available) OVERRIDE;
virtual int32_t SetSpeakerMute(bool enable) OVERRIDE;
virtual int32_t SpeakerMute(bool& enabled) const OVERRIDE;
// Microphone mute control
virtual int32_t MicrophoneMuteIsAvailable(bool& available) OVERRIDE;
virtual int32_t SetMicrophoneMute(bool enable) OVERRIDE;
virtual int32_t MicrophoneMute(bool& enabled) const OVERRIDE;
// Microphone boost control
virtual int32_t MicrophoneBoostIsAvailable(bool& available) OVERRIDE;
virtual int32_t SetMicrophoneBoost(bool enable) OVERRIDE;
virtual int32_t MicrophoneBoost(bool& enabled) const OVERRIDE;
// Stereo support
virtual int32_t StereoPlayoutIsAvailable(bool& available) OVERRIDE;
virtual int32_t SetStereoPlayout(bool enable) OVERRIDE;
virtual int32_t StereoPlayout(bool& enabled) const OVERRIDE;
virtual int32_t StereoRecordingIsAvailable(bool& available) OVERRIDE;
virtual int32_t SetStereoRecording(bool enable) OVERRIDE;
virtual int32_t StereoRecording(bool& enabled) const OVERRIDE;
// Delay information and control
virtual int32_t
SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
uint16_t sizeMS) OVERRIDE;
virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
uint16_t& sizeMS) const OVERRIDE;
virtual int32_t PlayoutDelay(uint16_t& delayMS) const OVERRIDE;
virtual int32_t RecordingDelay(uint16_t& delayMS) const OVERRIDE;
// CPU load
virtual int32_t CPULoad(uint16_t& load) const OVERRIDE;
virtual bool PlayoutWarning() const OVERRIDE;
virtual bool PlayoutError() const OVERRIDE;
virtual bool RecordingWarning() const OVERRIDE;
virtual bool RecordingError() const OVERRIDE;
virtual void ClearPlayoutWarning() OVERRIDE;
virtual void ClearPlayoutError() OVERRIDE;
virtual void ClearRecordingWarning() OVERRIDE;
virtual void ClearRecordingError() OVERRIDE;
virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) OVERRIDE;
void Lock() EXCLUSIVE_LOCK_FUNCTION(_critSect) {
void UnLock() UNLOCK_FUNCTION(_critSect) {
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 bool RecThreadFunc(void*);
static bool PlayThreadFunc(void*);
bool RecThreadProcess();
bool PlayThreadProcess();
AudioDeviceBuffer* _ptrAudioBuffer;
CriticalSectionWrapper& _critSect;
EventWrapper& _timeEventRec;
EventWrapper& _timeEventPlay;
EventWrapper& _recStartEvent;
EventWrapper& _playStartEvent;
ThreadWrapper* _ptrThreadPlay;
ThreadWrapper* _ptrThreadRec;
uint32_t _recThreadID;
uint32_t _playThreadID;
int32_t _id;
AudioMixerManagerLinuxPulse _mixerManager;
uint16_t _inputDeviceIndex;
uint16_t _outputDeviceIndex;
bool _inputDeviceIsSpecified;
bool _outputDeviceIsSpecified;
int sample_rate_hz_;
uint8_t _recChannels;
uint8_t _playChannels;
AudioDeviceModule::BufferType _playBufType;
bool _initialized;
bool _recording;
bool _playing;
bool _recIsInitialized;
bool _playIsInitialized;
bool _startRec;
bool _stopRec;
bool _startPlay;
bool _stopPlay;
bool _AGC;
bool update_speaker_volume_at_startup_;
uint16_t _playBufDelayFixed; // fixed playback delay
uint32_t _sndCardPlayDelay;
uint32_t _sndCardRecDelay;
int32_t _writeErrors;
uint16_t _playWarning;
uint16_t _playError;
uint16_t _recWarning;
uint16_t _recError;
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];
Display* _XDisplay;