audio_device_pulse_linux.h revision 12411ef40e08c5e28ccde54ab3418c96676ffcbc 
1/*
2 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 *
4 *  Use of this source code is governed by a BSD-style license
5 *  that can be found in the LICENSE file in the root of the source
6 *  tree. An additional intellectual property rights grant can be found
7 *  in the file PATENTS.  All contributing project authors may
8 *  be found in the AUTHORS file in the root of the source tree.
9 */
10
11#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
12#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
13
14#include "webrtc/base/platform_thread.h"
15#include "webrtc/base/thread_checker.h"
16#include "webrtc/modules/audio_device/audio_device_generic.h"
17#include "webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
18#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
19
20#include <X11/Xlib.h>
21#include <pulse/pulseaudio.h>
22
23// We define this flag if it's missing from our headers, because we want to be
24// able to compile against old headers but still use PA_STREAM_ADJUST_LATENCY
25// if run against a recent version of the library.
26#ifndef PA_STREAM_ADJUST_LATENCY
27#define PA_STREAM_ADJUST_LATENCY 0x2000U
28#endif
29#ifndef PA_STREAM_START_MUTED
30#define PA_STREAM_START_MUTED 0x1000U
31#endif
32
33// Set this constant to 0 to disable latency reading
34const uint32_t WEBRTC_PA_REPORT_LATENCY = 1;
35
36// Constants from implementation by Tristan Schmelcher [tschmelcher@google.com]
37
38// First PulseAudio protocol version that supports PA_STREAM_ADJUST_LATENCY.
39const uint32_t WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION = 13;
40
41// Some timing constants for optimal operation. See
42// https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008-January/001170.html
43// for a good explanation of some of the factors that go into this.
44
45// Playback.
46
47// For playback, there is a round-trip delay to fill the server-side playback
48// buffer, so setting too low of a latency is a buffer underflow risk. We will
49// automatically increase the latency if a buffer underflow does occur, but we
50// also enforce a sane minimum at start-up time. Anything lower would be
51// virtually guaranteed to underflow at least once, so there's no point in
52// allowing lower latencies.
53const uint32_t WEBRTC_PA_PLAYBACK_LATENCY_MINIMUM_MSECS = 20;
54
55// Every time a playback stream underflows, we will reconfigure it with target
56// latency that is greater by this amount.
57const uint32_t WEBRTC_PA_PLAYBACK_LATENCY_INCREMENT_MSECS = 20;
58
59// We also need to configure a suitable request size. Too small and we'd burn
60// CPU from the overhead of transfering small amounts of data at once. Too large
61// and the amount of data remaining in the buffer right before refilling it
62// would be a buffer underflow risk. We set it to half of the buffer size.
63const uint32_t WEBRTC_PA_PLAYBACK_REQUEST_FACTOR = 2;
64
65// Capture.
66
67// For capture, low latency is not a buffer overflow risk, but it makes us burn
68// CPU from the overhead of transfering small amounts of data at once, so we set
69// a recommended value that we use for the kLowLatency constant (but if the user
70// explicitly requests something lower then we will honour it).
71// 1ms takes about 6-7% CPU. 5ms takes about 5%. 10ms takes about 4.x%.
72const uint32_t WEBRTC_PA_LOW_CAPTURE_LATENCY_MSECS = 10;
73
74// There is a round-trip delay to ack the data to the server, so the
75// server-side buffer needs extra space to prevent buffer overflow. 20ms is
76// sufficient, but there is no penalty to making it bigger, so we make it huge.
77// (750ms is libpulse's default value for the _total_ buffer size in the
78// kNoLatencyRequirements case.)
79const uint32_t WEBRTC_PA_CAPTURE_BUFFER_EXTRA_MSECS = 750;
80
81const uint32_t WEBRTC_PA_MSECS_PER_SEC = 1000;
82
83// Init _configuredLatencyRec/Play to this value to disable latency requirements
84const int32_t WEBRTC_PA_NO_LATENCY_REQUIREMENTS = -1;
85
86// Set this const to 1 to account for peeked and used data in latency calculation
87const uint32_t WEBRTC_PA_CAPTURE_BUFFER_LATENCY_ADJUSTMENT = 0;
88
89namespace webrtc
90{
91class EventWrapper;
92
93class AudioDeviceLinuxPulse: public AudioDeviceGeneric
94{
95public:
96    AudioDeviceLinuxPulse(const int32_t id);
97    virtual ~AudioDeviceLinuxPulse();
98
99    // Retrieve the currently utilized audio layer
100    int32_t ActiveAudioLayer(
101        AudioDeviceModule::AudioLayer& audioLayer) const override;
102
103    // Main initializaton and termination
104    int32_t Init() override;
105    int32_t Terminate() override;
106    bool Initialized() const override;
107
108    // Device enumeration
109    int16_t PlayoutDevices() override;
110    int16_t RecordingDevices() override;
111    int32_t PlayoutDeviceName(uint16_t index,
112                              char name[kAdmMaxDeviceNameSize],
113                              char guid[kAdmMaxGuidSize]) override;
114    int32_t RecordingDeviceName(uint16_t index,
115                                char name[kAdmMaxDeviceNameSize],
116                                char guid[kAdmMaxGuidSize]) override;
117
118    // Device selection
119    int32_t SetPlayoutDevice(uint16_t index) override;
120    int32_t SetPlayoutDevice(
121        AudioDeviceModule::WindowsDeviceType device) override;
122    int32_t SetRecordingDevice(uint16_t index) override;
123    int32_t SetRecordingDevice(
124        AudioDeviceModule::WindowsDeviceType device) override;
125
126    // Audio transport initialization
127    int32_t PlayoutIsAvailable(bool& available) override;
128    int32_t InitPlayout() override;
129    bool PlayoutIsInitialized() const override;
130    int32_t RecordingIsAvailable(bool& available) override;
131    int32_t InitRecording() override;
132    bool RecordingIsInitialized() const override;
133
134    // Audio transport control
135    int32_t StartPlayout() override;
136    int32_t StopPlayout() override;
137    bool Playing() const override;
138    int32_t StartRecording() override;
139    int32_t StopRecording() override;
140    bool Recording() const override;
141
142    // Microphone Automatic Gain Control (AGC)
143    int32_t SetAGC(bool enable) override;
144    bool AGC() const override;
145
146    // Volume control based on the Windows Wave API (Windows only)
147    int32_t SetWaveOutVolume(uint16_t volumeLeft,
148                             uint16_t volumeRight) override;
149    int32_t WaveOutVolume(uint16_t& volumeLeft,
150                          uint16_t& volumeRight) const override;
151
152    // Audio mixer initialization
153    int32_t InitSpeaker() override;
154    bool SpeakerIsInitialized() const override;
155    int32_t InitMicrophone() override;
156    bool MicrophoneIsInitialized() const override;
157
158    // Speaker volume controls
159    int32_t SpeakerVolumeIsAvailable(bool& available) override;
160    int32_t SetSpeakerVolume(uint32_t volume) override;
161    int32_t SpeakerVolume(uint32_t& volume) const override;
162    int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
163    int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
164    int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const override;
165
166    // Microphone volume controls
167    int32_t MicrophoneVolumeIsAvailable(bool& available) override;
168    int32_t SetMicrophoneVolume(uint32_t volume) override;
169    int32_t MicrophoneVolume(uint32_t& volume) const override;
170    int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
171    int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
172    int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const override;
173
174    // Speaker mute control
175    int32_t SpeakerMuteIsAvailable(bool& available) override;
176    int32_t SetSpeakerMute(bool enable) override;
177    int32_t SpeakerMute(bool& enabled) const override;
178
179    // Microphone mute control
180    int32_t MicrophoneMuteIsAvailable(bool& available) override;
181    int32_t SetMicrophoneMute(bool enable) override;
182    int32_t MicrophoneMute(bool& enabled) const override;
183
184    // Microphone boost control
185    int32_t MicrophoneBoostIsAvailable(bool& available) override;
186    int32_t SetMicrophoneBoost(bool enable) override;
187    int32_t MicrophoneBoost(bool& enabled) const override;
188
189    // Stereo support
190    int32_t StereoPlayoutIsAvailable(bool& available) override;
191    int32_t SetStereoPlayout(bool enable) override;
192    int32_t StereoPlayout(bool& enabled) const override;
193    int32_t StereoRecordingIsAvailable(bool& available) override;
194    int32_t SetStereoRecording(bool enable) override;
195    int32_t StereoRecording(bool& enabled) const override;
196
197    // Delay information and control
198    int32_t SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
199                             uint16_t sizeMS) override;
200    int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
201                          uint16_t& sizeMS) const override;
202    int32_t PlayoutDelay(uint16_t& delayMS) const override;
203    int32_t RecordingDelay(uint16_t& delayMS) const override;
204
205    // CPU load
206    int32_t CPULoad(uint16_t& load) const override;
207
208    bool PlayoutWarning() const override;
209    bool PlayoutError() const override;
210    bool RecordingWarning() const override;
211    bool RecordingError() const override;
212    void ClearPlayoutWarning() override;
213    void ClearPlayoutError() override;
214    void ClearRecordingWarning() override;
215    void ClearRecordingError() override;
216
217   void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
218
219private:
220    void Lock() EXCLUSIVE_LOCK_FUNCTION(_critSect) {
221        _critSect.Enter();
222    }
223    void UnLock() UNLOCK_FUNCTION(_critSect) {
224        _critSect.Leave();
225    }
226    void WaitForOperationCompletion(pa_operation* paOperation) const;
227    void WaitForSuccess(pa_operation* paOperation) const;
228
229    bool KeyPressed() const;
230
231    static void PaContextStateCallback(pa_context *c, void *pThis);
232    static void PaSinkInfoCallback(pa_context *c, const pa_sink_info *i,
233                                   int eol, void *pThis);
234    static void PaSourceInfoCallback(pa_context *c, const pa_source_info *i,
235                                     int eol, void *pThis);
236    static void PaServerInfoCallback(pa_context *c, const pa_server_info *i,
237                                     void *pThis);
238    static void PaStreamStateCallback(pa_stream *p, void *pThis);
239    void PaContextStateCallbackHandler(pa_context *c);
240    void PaSinkInfoCallbackHandler(const pa_sink_info *i, int eol);
241    void PaSourceInfoCallbackHandler(const pa_source_info *i, int eol);
242    void PaServerInfoCallbackHandler(const pa_server_info *i);
243    void PaStreamStateCallbackHandler(pa_stream *p);
244
245    void EnableWriteCallback();
246    void DisableWriteCallback();
247    static void PaStreamWriteCallback(pa_stream *unused, size_t buffer_space,
248                                      void *pThis);
249    void PaStreamWriteCallbackHandler(size_t buffer_space);
250    static void PaStreamUnderflowCallback(pa_stream *unused, void *pThis);
251    void PaStreamUnderflowCallbackHandler();
252    void EnableReadCallback();
253    void DisableReadCallback();
254    static void PaStreamReadCallback(pa_stream *unused1, size_t unused2,
255                                     void *pThis);
256    void PaStreamReadCallbackHandler();
257    static void PaStreamOverflowCallback(pa_stream *unused, void *pThis);
258    void PaStreamOverflowCallbackHandler();
259    int32_t LatencyUsecs(pa_stream *stream);
260    int32_t ReadRecordedData(const void* bufferData, size_t bufferSize);
261    int32_t ProcessRecordedData(int8_t *bufferData,
262                                uint32_t bufferSizeInSamples,
263                                uint32_t recDelay);
264
265    int32_t CheckPulseAudioVersion();
266    int32_t InitSamplingFrequency();
267    int32_t GetDefaultDeviceInfo(bool recDevice, char* name, uint16_t& index);
268    int32_t InitPulseAudio();
269    int32_t TerminatePulseAudio();
270
271    void PaLock();
272    void PaUnLock();
273
274    static bool RecThreadFunc(void*);
275    static bool PlayThreadFunc(void*);
276    bool RecThreadProcess();
277    bool PlayThreadProcess();
278
279    AudioDeviceBuffer* _ptrAudioBuffer;
280
281    CriticalSectionWrapper& _critSect;
282    EventWrapper& _timeEventRec;
283    EventWrapper& _timeEventPlay;
284    EventWrapper& _recStartEvent;
285    EventWrapper& _playStartEvent;
286
287    rtc::scoped_ptr<PlatformThread> _ptrThreadPlay;
288    rtc::scoped_ptr<PlatformThread> _ptrThreadRec;
289    int32_t _id;
290
291    AudioMixerManagerLinuxPulse _mixerManager;
292
293    uint16_t _inputDeviceIndex;
294    uint16_t _outputDeviceIndex;
295    bool _inputDeviceIsSpecified;
296    bool _outputDeviceIsSpecified;
297
298    int sample_rate_hz_;
299    uint8_t _recChannels;
300    uint8_t _playChannels;
301
302    AudioDeviceModule::BufferType _playBufType;
303
304    // Stores thread ID in constructor.
305    // We can then use ThreadChecker::CalledOnValidThread() to ensure that
306    // other methods are called from the same thread.
307    // Currently only does RTC_DCHECK(thread_checker_.CalledOnValidThread()).
308    rtc::ThreadChecker thread_checker_;
309
310    bool _initialized;
311    bool _recording;
312    bool _playing;
313    bool _recIsInitialized;
314    bool _playIsInitialized;
315    bool _startRec;
316    bool _stopRec;
317    bool _startPlay;
318    bool _stopPlay;
319    bool _AGC;
320    bool update_speaker_volume_at_startup_;
321
322    uint16_t _playBufDelayFixed; // fixed playback delay
323
324    uint32_t _sndCardPlayDelay;
325    uint32_t _sndCardRecDelay;
326
327    int32_t _writeErrors;
328    uint16_t _playWarning;
329    uint16_t _playError;
330    uint16_t _recWarning;
331    uint16_t _recError;
332
333    uint16_t _deviceIndex;
334    int16_t _numPlayDevices;
335    int16_t _numRecDevices;
336    char* _playDeviceName;
337    char* _recDeviceName;
338    char* _playDisplayDeviceName;
339    char* _recDisplayDeviceName;
340    char _paServerVersion[32];
341
342    int8_t* _playBuffer;
343    size_t _playbackBufferSize;
344    size_t _playbackBufferUnused;
345    size_t _tempBufferSpace;
346    int8_t* _recBuffer;
347    size_t _recordBufferSize;
348    size_t _recordBufferUsed;
349    const void* _tempSampleData;
350    size_t _tempSampleDataSize;
351    int32_t _configuredLatencyPlay;
352    int32_t _configuredLatencyRec;
353
354    // PulseAudio
355    uint16_t _paDeviceIndex;
356    bool _paStateChanged;
357
358    pa_threaded_mainloop* _paMainloop;
359    pa_mainloop_api* _paMainloopApi;
360    pa_context* _paContext;
361
362    pa_stream* _recStream;
363    pa_stream* _playStream;
364    uint32_t _recStreamFlags;
365    uint32_t _playStreamFlags;
366    pa_buffer_attr _playBufferAttr;
367    pa_buffer_attr _recBufferAttr;
368
369    char _oldKeyState[32];
370    Display* _XDisplay;
371};
372
373}
374
375#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_PULSE_LINUX_H_
376