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* Copyright (c) 2011 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 "audio_device_generic.h"
#include "audio_mixer_manager_pulse_linux.h"
#include "critical_section_wrapper.h"
#include <pulse/pulseaudio.h>
// Set this define to make the code behave like in GTalk/libjingle
// 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 WebRtc_UWord32 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 WebRtc_UWord32 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 WebRtc_Word32 id);
static bool PulseAudioIsSupported();
// Retrieve the currently utilized audio layer
virtual WebRtc_Word32
ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
// Main initializaton and termination
virtual WebRtc_Word32 Init();
virtual WebRtc_Word32 Terminate();
virtual bool Initialized() const;
// Device enumeration
virtual WebRtc_Word16 PlayoutDevices();
virtual WebRtc_Word16 RecordingDevices();
virtual WebRtc_Word32 PlayoutDeviceName(
WebRtc_UWord16 index,
WebRtc_Word8 name[kAdmMaxDeviceNameSize],
WebRtc_Word8 guid[kAdmMaxGuidSize]);
virtual WebRtc_Word32 RecordingDeviceName(
WebRtc_UWord16 index,
WebRtc_Word8 name[kAdmMaxDeviceNameSize],
WebRtc_Word8 guid[kAdmMaxGuidSize]);
// Device selection
virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
virtual WebRtc_Word32 SetPlayoutDevice(
AudioDeviceModule::WindowsDeviceType device);
virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
virtual WebRtc_Word32 SetRecordingDevice(
AudioDeviceModule::WindowsDeviceType device);
// Audio transport initialization
virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
virtual WebRtc_Word32 InitPlayout();
virtual bool PlayoutIsInitialized() const;
virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
virtual WebRtc_Word32 InitRecording();
virtual bool RecordingIsInitialized() const;
// Audio transport control
virtual WebRtc_Word32 StartPlayout();
virtual WebRtc_Word32 StopPlayout();
virtual bool Playing() const;
virtual WebRtc_Word32 StartRecording();
virtual WebRtc_Word32 StopRecording();
virtual bool Recording() const;
// Microphone Automatic Gain Control (AGC)
virtual WebRtc_Word32 SetAGC(bool enable);
virtual bool AGC() const;
// Volume control based on the Windows Wave API (Windows only)
virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
WebRtc_UWord16 volumeRight);
virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft,
WebRtc_UWord16& volumeRight) const;
// Audio mixer initialization
virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
virtual WebRtc_Word32 InitSpeaker();
virtual bool SpeakerIsInitialized() const;
virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
virtual WebRtc_Word32 InitMicrophone();
virtual bool MicrophoneIsInitialized() const;
// Speaker volume controls
virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
// Microphone volume controls
virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
virtual WebRtc_Word32 MicrophoneVolumeStepSize(
WebRtc_UWord16& stepSize) const;
// Speaker mute control
virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
virtual WebRtc_Word32 SetSpeakerMute(bool enable);
virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
// Microphone mute control
virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
// Microphone boost control
virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
// Stereo support
virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
virtual WebRtc_Word32 SetStereoPlayout(bool enable);
virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
virtual WebRtc_Word32 SetStereoRecording(bool enable);
virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
// Delay information and control
virtual WebRtc_Word32
SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
WebRtc_UWord16 sizeMS);
virtual WebRtc_Word32 PlayoutBuffer(AudioDeviceModule::BufferType& type,
WebRtc_UWord16& sizeMS) const;
virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
// CPU load
virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
virtual bool PlayoutWarning() const;
virtual bool PlayoutError() const;
virtual bool RecordingWarning() const;
virtual bool RecordingError() const;
virtual void ClearPlayoutWarning();
virtual void ClearPlayoutError();
virtual void ClearRecordingWarning();
virtual void ClearRecordingError();
virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
void Lock()
void UnLock()
void WaitForOperationCompletion(pa_operation* paOperation) const;
void WaitForSuccess(pa_operation* paOperation) 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();
WebRtc_Word32 LatencyUsecs(pa_stream *stream);
WebRtc_Word32 ReadRecordedData(const void* bufferData, size_t bufferSize);
WebRtc_Word32 ProcessRecordedData(WebRtc_Word8 *bufferData,
WebRtc_UWord32 bufferSizeInSamples,
WebRtc_UWord32 recDelay);
WebRtc_Word32 CheckPulseAudioVersion();
WebRtc_Word32 InitSamplingFrequency();
WebRtc_Word32 GetDefaultDeviceInfo(bool recDevice, WebRtc_Word8* name,
WebRtc_UWord16& index);
WebRtc_Word32 InitPulseAudio();
WebRtc_Word32 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;
WebRtc_UWord32 _recThreadID;
WebRtc_UWord32 _playThreadID;
WebRtc_Word32 _id;
AudioMixerManagerLinuxPulse _mixerManager;
WebRtc_UWord16 _inputDeviceIndex;
WebRtc_UWord16 _outputDeviceIndex;
bool _inputDeviceIsSpecified;
bool _outputDeviceIsSpecified;
WebRtc_UWord32 _samplingFreq;
WebRtc_UWord8 _recChannels;
WebRtc_UWord8 _playChannels;
AudioDeviceModule::BufferType _playBufType;
bool _initialized;
bool _recording;
bool _playing;
bool _recIsInitialized;
bool _playIsInitialized;
bool _startRec;
bool _stopRec;
bool _startPlay;
bool _stopPlay;
bool _AGC;
WebRtc_UWord16 _playBufDelayFixed; // fixed playback delay
WebRtc_UWord32 _sndCardPlayDelay;
WebRtc_UWord32 _sndCardRecDelay;
WebRtc_Word32 _writeErrors;
WebRtc_UWord16 _playWarning;
WebRtc_UWord16 _playError;
WebRtc_UWord16 _recWarning;
WebRtc_UWord16 _recError;
WebRtc_UWord16 _deviceIndex;
WebRtc_Word16 _numPlayDevices;
WebRtc_Word16 _numRecDevices;
WebRtc_Word8* _playDeviceName;
WebRtc_Word8* _recDeviceName;
WebRtc_Word8* _playDisplayDeviceName;
WebRtc_Word8* _recDisplayDeviceName;
WebRtc_Word8 _paServerVersion[32];
WebRtc_Word8* _playBuffer;
size_t _playbackBufferSize;
size_t _playbackBufferUnused;
size_t _tempBufferSpace;
WebRtc_Word8* _recBuffer;
size_t _recordBufferSize;
size_t _recordBufferUsed;
const void* _tempSampleData;
size_t _tempSampleDataSize;
WebRtc_Word32 _configuredLatencyPlay;
WebRtc_Word32 _configuredLatencyRec;
// PulseAudio
WebRtc_UWord16 _paDeviceIndex;
bool _paStateChanged;
pa_threaded_mainloop* _paMainloop;
pa_mainloop_api* _paMainloopApi;
pa_context* _paContext;
pa_stream* _recStream;
pa_stream* _playStream;
WebRtc_UWord32 _recStreamFlags;
WebRtc_UWord32 _playStreamFlags;
pa_buffer_attr _playBufferAttr;
pa_buffer_attr _recBufferAttr;