delete old folders
Review URL: http://webrtc-codereview.appspot.com/105001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@348 4adac7df-926f-26a2-2b94-8c16560cd09d
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xians@google.com
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DUMMY_H
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#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DUMMY_H
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#include <stdio.h>
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#include "audio_device_generic.h"
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#include "critical_section_wrapper.h"
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namespace webrtc {
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class EventWrapper;
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class ThreadWrapper;
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class AudioDeviceDummy : public AudioDeviceGeneric
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{
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public:
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AudioDeviceDummy(const WebRtc_Word32 id);
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~AudioDeviceDummy();
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// Retrieve the currently utilized audio layer
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virtual WebRtc_Word32 ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
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// Main initializaton and termination
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virtual WebRtc_Word32 Init();
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virtual WebRtc_Word32 Terminate();
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virtual bool Initialized() const;
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// Device enumeration
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virtual WebRtc_Word16 PlayoutDevices();
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virtual WebRtc_Word16 RecordingDevices();
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virtual WebRtc_Word32 PlayoutDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize]);
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virtual WebRtc_Word32 RecordingDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize]);
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// Device selection
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virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
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virtual WebRtc_Word32 SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
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virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
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virtual WebRtc_Word32 SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device);
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// Audio transport initialization
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virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
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virtual WebRtc_Word32 InitPlayout();
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virtual bool PlayoutIsInitialized() const;
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virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
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virtual WebRtc_Word32 InitRecording();
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virtual bool RecordingIsInitialized() const;
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// Audio transport control
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virtual WebRtc_Word32 StartPlayout();
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virtual WebRtc_Word32 StopPlayout();
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virtual bool Playing() const;
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virtual WebRtc_Word32 StartRecording();
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virtual WebRtc_Word32 StopRecording();
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virtual bool Recording() const;
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// Microphone Automatic Gain Control (AGC)
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virtual WebRtc_Word32 SetAGC(bool enable);
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virtual bool AGC() const;
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// Volume control based on the Windows Wave API (Windows only)
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virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft, WebRtc_UWord16 volumeRight);
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virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft, WebRtc_UWord16& volumeRight) const;
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// Audio mixer initialization
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virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
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virtual WebRtc_Word32 InitSpeaker();
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virtual bool SpeakerIsInitialized() const;
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virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
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virtual WebRtc_Word32 InitMicrophone();
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virtual bool MicrophoneIsInitialized() const;
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// Speaker volume controls
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virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
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virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
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virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
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virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
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virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
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virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
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// Microphone volume controls
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virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
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virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
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virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
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virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
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virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
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virtual WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
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// Speaker mute control
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virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
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virtual WebRtc_Word32 SetSpeakerMute(bool enable);
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virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
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// Microphone mute control
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virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
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virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
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virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
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// Microphone boost control
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virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
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virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
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virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
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// Stereo support
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virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
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virtual WebRtc_Word32 SetStereoPlayout(bool enable);
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virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
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virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
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virtual WebRtc_Word32 SetStereoRecording(bool enable);
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virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
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// Delay information and control
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virtual WebRtc_Word32 SetPlayoutBuffer(const AudioDeviceModule::BufferType type, WebRtc_UWord16 sizeMS);
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virtual WebRtc_Word32 PlayoutBuffer(AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const;
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virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
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virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
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// CPU load
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virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
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virtual bool PlayoutWarning() const;
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virtual bool PlayoutError() const;
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virtual bool RecordingWarning() const;
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virtual bool RecordingError() const;
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virtual void ClearPlayoutWarning();
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virtual void ClearPlayoutError();
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virtual void ClearRecordingWarning();
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virtual void ClearRecordingError();
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virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
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private:
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void Lock() { _critSect.Enter(); };
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void UnLock() { _critSect.Leave(); };
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static bool RecThreadFunc(void*);
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static bool PlayThreadFunc(void*);
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bool RecThreadProcess();
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bool PlayThreadProcess();
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AudioDeviceBuffer* _ptrAudioBuffer;
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CriticalSectionWrapper& _critSect;
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WebRtc_Word32 _id;
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EventWrapper& _timeEventRec;
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EventWrapper& _timeEventPlay;
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EventWrapper& _recStartEvent;
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EventWrapper& _playStartEvent;
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ThreadWrapper* _ptrThreadRec;
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ThreadWrapper* _ptrThreadPlay;
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WebRtc_UWord32 _recThreadID;
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WebRtc_UWord32 _playThreadID;
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bool _initialized;
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bool _recording;
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bool _playing;
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bool _recIsInitialized;
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bool _playIsInitialized;
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bool _speakerIsInitialized;
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bool _microphoneIsInitialized;
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WebRtc_Word8 _recBuffer[2*160];
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FILE* _playDataFile;
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};
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DUMMY_H
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@ -1,58 +0,0 @@
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "audio_device_utility_dummy.h"
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#include "audio_device_config.h" // DEBUG_PRINT()
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#include "critical_section_wrapper.h"
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#include "trace.h"
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namespace webrtc
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{
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AudioDeviceUtilityDummy::AudioDeviceUtilityDummy(const WebRtc_Word32 id) :
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_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
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_id(id),
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_lastError(AudioDeviceModule::kAdmErrNone)
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{
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
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"%s created", __FUNCTION__);
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}
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AudioDeviceUtilityDummy::~AudioDeviceUtilityDummy()
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{
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
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"%s destroyed", __FUNCTION__);
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{
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CriticalSectionScoped lock(_critSect);
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// free stuff here...
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}
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delete &_critSect;
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}
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// ============================================================================
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// API
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// ============================================================================
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WebRtc_Word32 AudioDeviceUtilityDummy::Init()
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{
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WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
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"%s", __FUNCTION__);
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WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
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" OS info: %s", "Dummy");
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return 0;
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}
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} // namespace webrtc
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@ -1,37 +0,0 @@
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_DUMMY_H
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#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_DUMMY_H
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#include "audio_device_utility.h"
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#include "audio_device.h"
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namespace webrtc
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{
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class CriticalSectionWrapper;
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class AudioDeviceUtilityDummy: public AudioDeviceUtility
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{
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public:
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AudioDeviceUtilityDummy(const WebRtc_Word32 id);
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~AudioDeviceUtilityDummy();
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virtual WebRtc_Word32 Init();
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private:
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CriticalSectionWrapper& _critSect;
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WebRtc_Word32 _id;
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AudioDeviceModule::ErrorCode _lastError;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_UTILITY_DUMMY_H_
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@ -1,39 +0,0 @@
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/*
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* libjingle
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* Copyright 2004--2010, Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#include "alsasymboltable.h"
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namespace webrtc_adm_linux_alsa {
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LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(AlsaSymbolTable, "libasound.so.2")
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#define X(sym) \
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LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(AlsaSymbolTable, sym)
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ALSA_SYMBOLS_LIST
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#undef X
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LATE_BINDING_SYMBOL_TABLE_DEFINE_END(AlsaSymbolTable)
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} // namespace webrtc_adm_linux_alsa
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@ -1,144 +0,0 @@
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/*
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* libjingle
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* Copyright 2004--2010, Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
|
||||
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
|
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
|
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
|
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
|
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
|
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#ifndef WEBRTC_AUDIO_DEVICE_ALSASYMBOLTABLE_H
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#define WEBRTC_AUDIO_DEVICE_ALSASYMBOLTABLE_H
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#include "latebindingsymboltable.h"
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namespace webrtc_adm_linux_alsa {
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// The ALSA symbols we need, as an X-Macro list.
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// This list must contain precisely every libasound function that is used in
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// alsasoundsystem.cc.
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#define ALSA_SYMBOLS_LIST \
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X(snd_device_name_free_hint) \
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X(snd_device_name_get_hint) \
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X(snd_device_name_hint) \
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X(snd_pcm_avail_update) \
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X(snd_pcm_close) \
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X(snd_pcm_delay) \
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X(snd_pcm_drop) \
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X(snd_pcm_open) \
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X(snd_pcm_prepare) \
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X(snd_pcm_readi) \
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X(snd_pcm_recover) \
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X(snd_pcm_resume) \
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X(snd_pcm_reset) \
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X(snd_pcm_state) \
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X(snd_pcm_set_params) \
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X(snd_pcm_start) \
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X(snd_pcm_stream) \
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X(snd_pcm_wait) \
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X(snd_pcm_writei) \
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X(snd_pcm_info_get_class) \
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X(snd_pcm_info_get_subdevices_avail) \
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X(snd_pcm_info_get_subdevice_name) \
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X(snd_pcm_info_set_subdevice) \
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X(snd_pcm_info_get_id) \
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X(snd_pcm_info_set_device) \
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X(snd_pcm_info_set_stream) \
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X(snd_pcm_info_get_name) \
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X(snd_pcm_info_get_subdevices_count) \
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X(snd_pcm_info_sizeof) \
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X(snd_pcm_hw_params) \
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X(snd_pcm_hw_params_malloc) \
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X(snd_pcm_hw_params_free) \
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X(snd_pcm_hw_params_any) \
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X(snd_pcm_hw_params_set_access) \
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X(snd_pcm_hw_params_set_format) \
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X(snd_pcm_hw_params_set_channels) \
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X(snd_pcm_hw_params_set_rate_near) \
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X(snd_pcm_hw_params_set_buffer_size_near) \
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X(snd_card_next) \
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X(snd_card_get_name) \
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X(snd_config_update) \
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X(snd_config_copy) \
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X(snd_config_get_id) \
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X(snd_ctl_open) \
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X(snd_ctl_close) \
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X(snd_ctl_card_info) \
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X(snd_ctl_card_info_sizeof) \
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X(snd_ctl_card_info_get_id) \
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X(snd_ctl_card_info_get_name) \
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X(snd_ctl_pcm_next_device) \
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X(snd_ctl_pcm_info) \
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X(snd_mixer_load) \
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X(snd_mixer_free) \
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X(snd_mixer_detach) \
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X(snd_mixer_close) \
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X(snd_mixer_open) \
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X(snd_mixer_attach) \
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X(snd_mixer_first_elem) \
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X(snd_mixer_elem_next) \
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X(snd_mixer_selem_get_name) \
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X(snd_mixer_selem_is_active) \
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X(snd_mixer_selem_register) \
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X(snd_mixer_selem_set_playback_volume_all) \
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X(snd_mixer_selem_get_playback_volume) \
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X(snd_mixer_selem_has_playback_volume) \
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X(snd_mixer_selem_get_playback_volume_range) \
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X(snd_mixer_selem_has_playback_switch) \
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X(snd_mixer_selem_get_playback_switch) \
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X(snd_mixer_selem_set_playback_switch_all) \
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X(snd_mixer_selem_has_capture_switch) \
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X(snd_mixer_selem_get_capture_switch) \
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X(snd_mixer_selem_set_capture_switch_all) \
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X(snd_mixer_selem_has_capture_volume) \
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X(snd_mixer_selem_set_capture_volume_all) \
|
||||
X(snd_mixer_selem_get_capture_volume) \
|
||||
X(snd_mixer_selem_get_capture_volume_range) \
|
||||
X(snd_dlopen) \
|
||||
X(snd_dlclose) \
|
||||
X(snd_config) \
|
||||
X(snd_config_search) \
|
||||
X(snd_config_get_string) \
|
||||
X(snd_config_search_definition) \
|
||||
X(snd_config_get_type) \
|
||||
X(snd_config_delete) \
|
||||
X(snd_config_iterator_entry) \
|
||||
X(snd_config_iterator_first) \
|
||||
X(snd_config_iterator_next) \
|
||||
X(snd_config_iterator_end) \
|
||||
X(snd_config_delete_compound_members) \
|
||||
X(snd_config_get_integer) \
|
||||
X(snd_config_get_bool) \
|
||||
X(snd_dlsym) \
|
||||
X(snd_strerror) \
|
||||
X(snd_lib_error) \
|
||||
X(snd_lib_error_set_handler)
|
||||
|
||||
LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(AlsaSymbolTable)
|
||||
#define X(sym) \
|
||||
LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(AlsaSymbolTable, sym)
|
||||
ALSA_SYMBOLS_LIST
|
||||
#undef X
|
||||
LATE_BINDING_SYMBOL_TABLE_DECLARE_END(AlsaSymbolTable)
|
||||
|
||||
} // namespace webrtc_adm_linux_alsa
|
||||
|
||||
#endif // WEBRTC_AUDIO_DEVICE_ALSASYMBOLTABLE_H
|
||||
File diff suppressed because it is too large
Load Diff
@ -1,284 +0,0 @@
|
||||
/*
|
||||
* 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.
|
||||
*/
|
||||
|
||||
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_LINUX_ALSA_H
|
||||
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_LINUX_ALSA_H
|
||||
|
||||
#include "audio_device_generic.h"
|
||||
#include "critical_section_wrapper.h"
|
||||
#include "audio_mixer_manager_linux_alsa.h"
|
||||
|
||||
#include <sys/soundcard.h>
|
||||
#include <sys/ioctl.h>
|
||||
|
||||
#include <alsa/asoundlib.h>
|
||||
|
||||
namespace webrtc
|
||||
{
|
||||
class EventWrapper;
|
||||
class ThreadWrapper;
|
||||
|
||||
// Number of continuous buffer check errors before going 0->1
|
||||
const WebRtc_UWord16 THR_OLD_BUFFER_CHECK_METHOD = 30;
|
||||
// Number of buffer check errors before going 1->2
|
||||
const WebRtc_UWord16 THR_IGNORE_BUFFER_CHECK = 30;
|
||||
// 2.7 seconds (decimal 131071)
|
||||
const WebRtc_UWord32 ALSA_SNDCARD_BUFF_SIZE_REC = 0x1ffff;
|
||||
// ~170 ms (decimal 8191) - enough since we only write to buffer if it contains
|
||||
// less than 50 ms
|
||||
const WebRtc_UWord32 ALSA_SNDCARD_BUFF_SIZE_PLAY = 0x1fff;
|
||||
|
||||
const WebRtc_UWord32 REC_TIMER_PERIOD_MS = 2;
|
||||
const WebRtc_UWord32 PLAY_TIMER_PERIOD_MS = 5;
|
||||
const WebRtc_UWord16 PLAYBACK_THRESHOLD = 50;
|
||||
|
||||
const WebRtc_UWord32 REC_SAMPLES_PER_MS = 48;
|
||||
const WebRtc_UWord32 PLAY_SAMPLES_PER_MS = 48;
|
||||
|
||||
class AudioDeviceLinuxALSA : public AudioDeviceGeneric
|
||||
{
|
||||
public:
|
||||
AudioDeviceLinuxALSA(const WebRtc_Word32 id);
|
||||
~AudioDeviceLinuxALSA();
|
||||
|
||||
// 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;
|
||||
|
||||
public:
|
||||
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();
|
||||
|
||||
public:
|
||||
virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
|
||||
|
||||
private:
|
||||
WebRtc_Word32 GetDevicesInfo(const WebRtc_Word32 function,
|
||||
const bool playback,
|
||||
const WebRtc_Word32 enumDeviceNo = 0,
|
||||
char* enumDeviceName = NULL,
|
||||
const WebRtc_Word32 ednLen = 0) const;
|
||||
WebRtc_Word32 ErrorRecovery(WebRtc_Word32 error, snd_pcm_t* deviceHandle);
|
||||
void FillPlayoutBuffer();
|
||||
|
||||
private:
|
||||
void Lock() { _critSect.Enter(); };
|
||||
void UnLock() { _critSect.Leave(); };
|
||||
private:
|
||||
inline WebRtc_Word32 InputSanityCheckAfterUnlockedPeriod() const;
|
||||
inline WebRtc_Word32 OutputSanityCheckAfterUnlockedPeriod() const;
|
||||
|
||||
WebRtc_Word32 PrepareStartRecording();
|
||||
WebRtc_Word32 GetPlayoutBufferDelay();
|
||||
WebRtc_Word32 GetRecordingBufferDelay(bool preRead);
|
||||
|
||||
private:
|
||||
static bool RecThreadFunc(void*);
|
||||
static bool PlayThreadFunc(void*);
|
||||
bool RecThreadProcess();
|
||||
bool PlayThreadProcess();
|
||||
|
||||
private:
|
||||
AudioDeviceBuffer* _ptrAudioBuffer;
|
||||
|
||||
CriticalSectionWrapper& _critSect;
|
||||
EventWrapper& _timeEventRec;
|
||||
EventWrapper& _timeEventPlay;
|
||||
EventWrapper& _recStartEvent;
|
||||
EventWrapper& _playStartEvent;
|
||||
|
||||
ThreadWrapper* _ptrThreadRec;
|
||||
ThreadWrapper* _ptrThreadPlay;
|
||||
WebRtc_UWord32 _recThreadID;
|
||||
WebRtc_UWord32 _playThreadID;
|
||||
|
||||
WebRtc_Word32 _id;
|
||||
|
||||
AudioMixerManagerLinuxALSA _mixerManager;
|
||||
|
||||
WebRtc_UWord16 _inputDeviceIndex;
|
||||
WebRtc_UWord16 _outputDeviceIndex;
|
||||
bool _inputDeviceIsSpecified;
|
||||
bool _outputDeviceIsSpecified;
|
||||
|
||||
snd_pcm_t* _handleRecord;
|
||||
snd_pcm_t* _handlePlayout;
|
||||
|
||||
snd_pcm_uframes_t _recSndcardBuffsize;
|
||||
snd_pcm_uframes_t _playSndcardBuffsize;
|
||||
|
||||
WebRtc_UWord32 _samplingFreqRec;
|
||||
WebRtc_UWord32 _samplingFreqPlay;
|
||||
WebRtc_UWord8 _recChannels;
|
||||
WebRtc_UWord8 _playChannels;
|
||||
|
||||
WebRtc_UWord32 _playbackBufferSize;
|
||||
WebRtc_UWord32 _recordBufferSize;
|
||||
WebRtc_Word16* _recBuffer;
|
||||
AudioDeviceModule::BufferType _playBufType;
|
||||
|
||||
private:
|
||||
bool _initialized;
|
||||
bool _recording;
|
||||
bool _playing;
|
||||
bool _recIsInitialized;
|
||||
bool _playIsInitialized;
|
||||
bool _startRec;
|
||||
bool _stopRec;
|
||||
bool _startPlay;
|
||||
bool _stopPlay;
|
||||
bool _AGC;
|
||||
bool _buffersizeFromZeroAvail;
|
||||
bool _buffersizeFromZeroDelay;
|
||||
|
||||
WebRtc_UWord32 _sndCardPlayDelay; // Just to store last value
|
||||
WebRtc_UWord32 _previousSndCardPlayDelay; // Stores previous _sndCardPlayDelay value
|
||||
WebRtc_UWord8 _delayMonitorStatePlay; // 0 normal, 1 monitor delay change (after error)
|
||||
WebRtc_Word16 _largeDelayCountPlay; // Used when monitoring delay change
|
||||
WebRtc_UWord32 _sndCardRecDelay;
|
||||
WebRtc_UWord32 _numReadyRecSamples;
|
||||
|
||||
WebRtc_UWord8 _bufferCheckMethodPlay;
|
||||
WebRtc_UWord8 _bufferCheckMethodRec;
|
||||
WebRtc_UWord32 _bufferCheckErrorsPlay;
|
||||
WebRtc_UWord32 _bufferCheckErrorsRec;
|
||||
WebRtc_Word32 _lastBufferCheckValuePlay;
|
||||
WebRtc_Word32 _writeErrors;
|
||||
|
||||
WebRtc_UWord16 _playWarning;
|
||||
WebRtc_UWord16 _playError;
|
||||
WebRtc_UWord16 _recWarning;
|
||||
WebRtc_UWord16 _recError;
|
||||
|
||||
WebRtc_UWord16 _playBufDelay; // playback delay
|
||||
WebRtc_UWord16 _playBufDelayFixed; // fixed playback delay
|
||||
};
|
||||
|
||||
}
|
||||
|
||||
#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_LINUX_ALSA_H_
|
||||
File diff suppressed because it is too large
Load Diff
@ -1,385 +0,0 @@
|
||||
/*
|
||||
* 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.
|
||||
*/
|
||||
|
||||
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_LINUX_PULSE_H
|
||||
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_LINUX_PULSE_H
|
||||
|
||||
#include "audio_device_generic.h"
|
||||
#include "audio_mixer_manager_linux_pulse.h"
|
||||
#include "critical_section_wrapper.h"
|
||||
|
||||
#include <pulse/pulseaudio.h>
|
||||
|
||||
// Set this define to make the code behave like in GTalk/libjingle
|
||||
//#define WEBRTC_PA_GTALK
|
||||
|
||||
// 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 WebRtc_UWord32 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 WebRtc_UWord32 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 WebRtc_UWord32 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 WebRtc_UWord32 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 WebRtc_UWord32 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 WebRtc_UWord32 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 WebRtc_UWord32 WEBRTC_PA_CAPTURE_BUFFER_EXTRA_MSECS = 750;
|
||||
|
||||
const WebRtc_UWord32 WEBRTC_PA_MSECS_PER_SEC = 1000;
|
||||
|
||||
// Init _configuredLatencyRec/Play to this value to disable latency requirements
|
||||
const WebRtc_Word32 WEBRTC_PA_NO_LATENCY_REQUIREMENTS = -1;
|
||||
|
||||
// Set this const to 1 to account for peeked and used data in latency calculation
|
||||
const WebRtc_UWord32 WEBRTC_PA_CAPTURE_BUFFER_LATENCY_ADJUSTMENT = 0;
|
||||
|
||||
namespace webrtc
|
||||
{
|
||||
class EventWrapper;
|
||||
class ThreadWrapper;
|
||||
|
||||
class AudioDeviceLinuxPulse: public AudioDeviceGeneric
|
||||
{
|
||||
public:
|
||||
AudioDeviceLinuxPulse(const WebRtc_Word32 id);
|
||||
~AudioDeviceLinuxPulse();
|
||||
|
||||
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;
|
||||
|
||||
public:
|
||||
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();
|
||||
|
||||
public:
|
||||
virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
|
||||
|
||||
private:
|
||||
void Lock()
|
||||
{
|
||||
_critSect.Enter();
|
||||
}
|
||||
;
|
||||
void UnLock()
|
||||
{
|
||||
_critSect.Leave();
|
||||
}
|
||||
;
|
||||
void WaitForOperationCompletion(pa_operation* paOperation) const;
|
||||
void WaitForSuccess(pa_operation* paOperation) const;
|
||||
|
||||
private:
|
||||
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();
|
||||
|
||||
private:
|
||||
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;
|
||||
|
||||
private:
|
||||
bool _initialized;
|
||||
bool _recording;
|
||||
bool _playing;
|
||||
bool _recIsInitialized;
|
||||
bool _playIsInitialized;
|
||||
bool _startRec;
|
||||
bool _stopRec;
|
||||
bool _startPlay;
|
||||
bool _stopPlay;
|
||||
bool _AGC;
|
||||
|
||||
private:
|
||||
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;
|
||||
};
|
||||
|
||||
}
|
||||
|
||||
#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_LINUX_PULSE_H_
|
||||
@ -1,57 +0,0 @@
|
||||
/*
|
||||
* 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_utility_linux.h"
|
||||
#include "audio_device_config.h" // DEBUG_PRINT()
|
||||
#include "critical_section_wrapper.h"
|
||||
#include "trace.h"
|
||||
|
||||
namespace webrtc
|
||||
{
|
||||
|
||||
AudioDeviceUtilityLinux::AudioDeviceUtilityLinux(const WebRtc_Word32 id) :
|
||||
_critSect(*CriticalSectionWrapper::CreateCriticalSection()), _id(id),
|
||||
_lastError(AudioDeviceModule::kAdmErrNone)
|
||||
{
|
||||
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
|
||||
"%s created", __FUNCTION__);
|
||||
}
|
||||
|
||||
AudioDeviceUtilityLinux::~AudioDeviceUtilityLinux()
|
||||
{
|
||||
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
|
||||
"%s destroyed", __FUNCTION__);
|
||||
{
|
||||
CriticalSectionScoped lock(_critSect);
|
||||
|
||||
// free stuff here...
|
||||
}
|
||||
|
||||
delete &_critSect;
|
||||
}
|
||||
|
||||
// ============================================================================
|
||||
// API
|
||||
// ============================================================================
|
||||
|
||||
|
||||
WebRtc_Word32 AudioDeviceUtilityLinux::Init()
|
||||
{
|
||||
WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
|
||||
"%s", __FUNCTION__);
|
||||
|
||||
WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
|
||||
" OS info: %s", "Linux");
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
||||
} // namespace webrtc
|
||||
@ -1,37 +0,0 @@
|
||||
/*
|
||||
* 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.
|
||||
*/
|
||||
|
||||
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_LINUX_H
|
||||
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_LINUX_H
|
||||
|
||||
#include "audio_device_utility.h"
|
||||
#include "audio_device.h"
|
||||
|
||||
namespace webrtc
|
||||
{
|
||||
class CriticalSectionWrapper;
|
||||
|
||||
class AudioDeviceUtilityLinux: public AudioDeviceUtility
|
||||
{
|
||||
public:
|
||||
AudioDeviceUtilityLinux(const WebRtc_Word32 id);
|
||||
~AudioDeviceUtilityLinux();
|
||||
|
||||
virtual WebRtc_Word32 Init();
|
||||
|
||||
private:
|
||||
CriticalSectionWrapper& _critSect;
|
||||
WebRtc_Word32 _id;
|
||||
AudioDeviceModule::ErrorCode _lastError;
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_UTILITY_LINUX_H_
|
||||
File diff suppressed because it is too large
Load Diff
@ -1,78 +0,0 @@
|
||||
/*
|
||||
* 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.
|
||||
*/
|
||||
|
||||
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_LINUX_ALSA_H
|
||||
#define WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_LINUX_ALSA_H
|
||||
|
||||
#include "typedefs.h"
|
||||
#include "audio_device.h"
|
||||
#include "critical_section_wrapper.h"
|
||||
#include "alsasymboltable.h"
|
||||
|
||||
#include <alsa/asoundlib.h>
|
||||
|
||||
namespace webrtc
|
||||
{
|
||||
|
||||
class AudioMixerManagerLinuxALSA
|
||||
{
|
||||
public:
|
||||
WebRtc_Word32 OpenSpeaker(char* deviceName);
|
||||
WebRtc_Word32 OpenMicrophone(char* deviceName);
|
||||
WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
|
||||
WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
|
||||
WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
|
||||
WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
|
||||
WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
|
||||
WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
|
||||
WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
|
||||
WebRtc_Word32 SetSpeakerMute(bool enable);
|
||||
WebRtc_Word32 SpeakerMute(bool& enabled) const;
|
||||
WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
|
||||
WebRtc_Word32 SetMicrophoneMute(bool enable);
|
||||
WebRtc_Word32 MicrophoneMute(bool& enabled) const;
|
||||
WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
|
||||
WebRtc_Word32 SetMicrophoneBoost(bool enable);
|
||||
WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
|
||||
WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
|
||||
WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
|
||||
WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
|
||||
WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
|
||||
WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
|
||||
WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
|
||||
WebRtc_Word32 Close();
|
||||
WebRtc_Word32 CloseSpeaker();
|
||||
WebRtc_Word32 CloseMicrophone();
|
||||
bool SpeakerIsInitialized() const;
|
||||
bool MicrophoneIsInitialized() const;
|
||||
|
||||
public:
|
||||
AudioMixerManagerLinuxALSA(const WebRtc_Word32 id);
|
||||
~AudioMixerManagerLinuxALSA();
|
||||
|
||||
private:
|
||||
WebRtc_Word32 LoadMicMixerElement() const;
|
||||
WebRtc_Word32 LoadSpeakerMixerElement() const;
|
||||
void GetControlName(char *controlName, char* deviceName) const;
|
||||
|
||||
private:
|
||||
CriticalSectionWrapper& _critSect;
|
||||
WebRtc_Word32 _id;
|
||||
mutable snd_mixer_t* _outputMixerHandle;
|
||||
char _outputMixerStr[kAdmMaxDeviceNameSize];
|
||||
mutable snd_mixer_t* _inputMixerHandle;
|
||||
char _inputMixerStr[kAdmMaxDeviceNameSize];
|
||||
mutable snd_mixer_elem_t* _outputMixerElement;
|
||||
mutable snd_mixer_elem_t* _inputMixerElement;
|
||||
};
|
||||
|
||||
}
|
||||
|
||||
#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_MIXER_MANAGER_LINUX_ALSA_H_
|
||||
File diff suppressed because it is too large
Load Diff
@ -1,117 +0,0 @@
|
||||
/*
|
||||
* 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.
|
||||
*/
|
||||
|
||||
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_LINUX_PULSE_H
|
||||
#define WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_LINUX_PULSE_H
|
||||
|
||||
#include "typedefs.h"
|
||||
#include "audio_device.h"
|
||||
#include "critical_section_wrapper.h"
|
||||
#include "pulseaudiosymboltable.h"
|
||||
|
||||
#include <stdint.h>
|
||||
#include <pulse/pulseaudio.h>
|
||||
|
||||
#ifndef UINT32_MAX
|
||||
#define UINT32_MAX ((uint32_t)-1)
|
||||
#endif
|
||||
|
||||
namespace webrtc
|
||||
{
|
||||
|
||||
class AudioMixerManagerLinuxPulse
|
||||
{
|
||||
public:
|
||||
WebRtc_Word32 SetPlayStream(pa_stream* playStream);
|
||||
WebRtc_Word32 SetRecStream(pa_stream* recStream);
|
||||
WebRtc_Word32 OpenSpeaker(WebRtc_UWord16 deviceIndex);
|
||||
WebRtc_Word32 OpenMicrophone(WebRtc_UWord16 deviceIndex);
|
||||
WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
|
||||
WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
|
||||
WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
|
||||
WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
|
||||
WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
|
||||
WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
|
||||
WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
|
||||
WebRtc_Word32 SetSpeakerMute(bool enable);
|
||||
WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
|
||||
WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
|
||||
WebRtc_Word32 SpeakerMute(bool& enabled) const;
|
||||
WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
|
||||
WebRtc_Word32 SetMicrophoneMute(bool enable);
|
||||
WebRtc_Word32 MicrophoneMute(bool& enabled) const;
|
||||
WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
|
||||
WebRtc_Word32 SetMicrophoneBoost(bool enable);
|
||||
WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
|
||||
WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
|
||||
WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
|
||||
WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
|
||||
WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
|
||||
WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
|
||||
WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
|
||||
WebRtc_Word32 SetPulseAudioObjects(pa_threaded_mainloop* mainloop,
|
||||
pa_context* context);
|
||||
WebRtc_Word32 Close();
|
||||
WebRtc_Word32 CloseSpeaker();
|
||||
WebRtc_Word32 CloseMicrophone();
|
||||
bool SpeakerIsInitialized() const;
|
||||
bool MicrophoneIsInitialized() const;
|
||||
|
||||
public:
|
||||
AudioMixerManagerLinuxPulse(const WebRtc_Word32 id);
|
||||
~AudioMixerManagerLinuxPulse();
|
||||
|
||||
private:
|
||||
static void PaSinkInfoCallback(pa_context *c, const pa_sink_info *i,
|
||||
int eol, void *pThis);
|
||||
static void PaSinkInputInfoCallback(pa_context *c,
|
||||
const pa_sink_input_info *i, int eol,
|
||||
void *pThis);
|
||||
static void PaSourceInfoCallback(pa_context *c, const pa_source_info *i,
|
||||
int eol, void *pThis);
|
||||
static void
|
||||
PaSetVolumeCallback(pa_context * /*c*/, int success, void */*pThis*/);
|
||||
void PaSinkInfoCallbackHandler(const pa_sink_info *i, int eol);
|
||||
void PaSinkInputInfoCallbackHandler(const pa_sink_input_info *i, int eol);
|
||||
void PaSourceInfoCallbackHandler(const pa_source_info *i, int eol);
|
||||
|
||||
void ResetCallbackVariables() const;
|
||||
void WaitForOperationCompletion(pa_operation* paOperation) const;
|
||||
void PaLock() const;
|
||||
void PaUnLock() const;
|
||||
|
||||
private:
|
||||
CriticalSectionWrapper& _critSect;
|
||||
WebRtc_Word32 _id;
|
||||
WebRtc_Word16 _paOutputDeviceIndex;
|
||||
WebRtc_Word16 _paInputDeviceIndex;
|
||||
|
||||
pa_stream* _paPlayStream;
|
||||
pa_stream* _paRecStream;
|
||||
|
||||
pa_threaded_mainloop* _paMainloop;
|
||||
pa_context* _paContext;
|
||||
|
||||
mutable WebRtc_UWord32 _paVolume;
|
||||
mutable WebRtc_UWord32 _paMute;
|
||||
mutable WebRtc_UWord32 _paVolSteps;
|
||||
bool _paSpeakerMute;
|
||||
mutable WebRtc_UWord32 _paSpeakerVolume;
|
||||
mutable WebRtc_UWord8 _paChannels;
|
||||
bool _paObjectsSet;
|
||||
mutable bool _callbackValues;
|
||||
|
||||
WebRtc_UWord8 _micVolChannels;
|
||||
WebRtc_UWord8 _spkVolChannels;
|
||||
};
|
||||
|
||||
}
|
||||
|
||||
#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_MIXER_MANAGER_LINUX_PULSE_H_
|
||||
@ -1,116 +0,0 @@
|
||||
/*
|
||||
* libjingle
|
||||
* Copyright 2004--2010, Google Inc.
|
||||
*
|
||||
* Redistribution and use in source and binary forms, with or without
|
||||
* modification, are permitted provided that the following conditions are met:
|
||||
*
|
||||
* 1. Redistributions of source code must retain the above copyright notice,
|
||||
* this list of conditions and the following disclaimer.
|
||||
* 2. Redistributions in binary form must reproduce the above copyright notice,
|
||||
* this list of conditions and the following disclaimer in the documentation
|
||||
* and/or other materials provided with the distribution.
|
||||
* 3. The name of the author may not be used to endorse or promote products
|
||||
* derived from this software without specific prior written permission.
|
||||
*
|
||||
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
|
||||
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
|
||||
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
|
||||
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
|
||||
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
|
||||
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
|
||||
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
|
||||
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
|
||||
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
|
||||
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
||||
*/
|
||||
|
||||
#include "latebindingsymboltable.h"
|
||||
|
||||
#ifdef WEBRTC_LINUX
|
||||
#include <dlfcn.h>
|
||||
#endif
|
||||
|
||||
// TODO(grunell): Either put inside webrtc namespace or use webrtc:: instead.
|
||||
using namespace webrtc;
|
||||
|
||||
namespace webrtc_adm_linux {
|
||||
|
||||
inline static const char *GetDllError() {
|
||||
#ifdef WEBRTC_LINUX
|
||||
char *err = dlerror();
|
||||
if (err) {
|
||||
return err;
|
||||
} else {
|
||||
return "No error";
|
||||
}
|
||||
#else
|
||||
#error Not implemented
|
||||
#endif
|
||||
}
|
||||
|
||||
DllHandle InternalLoadDll(const char dll_name[]) {
|
||||
#ifdef WEBRTC_LINUX
|
||||
DllHandle handle = dlopen(dll_name, RTLD_NOW);
|
||||
#else
|
||||
#error Not implemented
|
||||
#endif
|
||||
if (handle == kInvalidDllHandle) {
|
||||
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, -1,
|
||||
"Can't load %s : %d", dll_name, GetDllError());
|
||||
}
|
||||
return handle;
|
||||
}
|
||||
|
||||
void InternalUnloadDll(DllHandle handle) {
|
||||
#ifdef WEBRTC_LINUX
|
||||
if (dlclose(handle) != 0) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
|
||||
"%d", GetDllError());
|
||||
}
|
||||
#else
|
||||
#error Not implemented
|
||||
#endif
|
||||
}
|
||||
|
||||
static bool LoadSymbol(DllHandle handle,
|
||||
const char *symbol_name,
|
||||
void **symbol) {
|
||||
#ifdef WEBRTC_LINUX
|
||||
*symbol = dlsym(handle, symbol_name);
|
||||
char *err = dlerror();
|
||||
if (err) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
|
||||
"Error loading symbol %s : %d", symbol_name, err);
|
||||
return false;
|
||||
} else if (!*symbol) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
|
||||
"Symbol %s is NULL", symbol_name);
|
||||
return false;
|
||||
}
|
||||
return true;
|
||||
#else
|
||||
#error Not implemented
|
||||
#endif
|
||||
}
|
||||
|
||||
// This routine MUST assign SOME value for every symbol, even if that value is
|
||||
// NULL, or else some symbols may be left with uninitialized data that the
|
||||
// caller may later interpret as a valid address.
|
||||
bool InternalLoadSymbols(DllHandle handle,
|
||||
int num_symbols,
|
||||
const char *const symbol_names[],
|
||||
void *symbols[]) {
|
||||
#ifdef WEBRTC_LINUX
|
||||
// Clear any old errors.
|
||||
dlerror();
|
||||
#endif
|
||||
for (int i = 0; i < num_symbols; ++i) {
|
||||
if (!LoadSymbol(handle, symbol_names[i], &symbols[i])) {
|
||||
return false;
|
||||
}
|
||||
}
|
||||
return true;
|
||||
}
|
||||
|
||||
} // namespace webrtc_adm_linux
|
||||
@ -1,195 +0,0 @@
|
||||
/*
|
||||
* libjingle
|
||||
* Copyright 2004--2010, Google Inc.
|
||||
*
|
||||
* Redistribution and use in source and binary forms, with or without
|
||||
* modification, are permitted provided that the following conditions are met:
|
||||
*
|
||||
* 1. Redistributions of source code must retain the above copyright notice,
|
||||
* this list of conditions and the following disclaimer.
|
||||
* 2. Redistributions in binary form must reproduce the above copyright notice,
|
||||
* this list of conditions and the following disclaimer in the documentation
|
||||
* and/or other materials provided with the distribution.
|
||||
* 3. The name of the author may not be used to endorse or promote products
|
||||
* derived from this software without specific prior written permission.
|
||||
*
|
||||
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
|
||||
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
|
||||
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
|
||||
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
|
||||
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
|
||||
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
|
||||
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
|
||||
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
|
||||
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
|
||||
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
||||
*/
|
||||
|
||||
#ifndef WEBRTC_AUDIO_DEVICE_LATEBINDINGSYMBOLTABLE_H
|
||||
#define WEBRTC_AUDIO_DEVICE_LATEBINDINGSYMBOLTABLE_H
|
||||
|
||||
#include <assert.h>
|
||||
#include <stddef.h> // for NULL
|
||||
#include <string.h>
|
||||
|
||||
#include "constructor_magic.h"
|
||||
#include "trace.h"
|
||||
|
||||
// This file provides macros for creating "symbol table" classes to simplify the
|
||||
// dynamic loading of symbols from DLLs. Currently the implementation only
|
||||
// supports Linux and pure C symbols.
|
||||
// See talk/sound/pulseaudiosymboltable.(h|cc) for an example.
|
||||
|
||||
namespace webrtc_adm_linux {
|
||||
|
||||
#ifdef WEBRTC_LINUX
|
||||
typedef void *DllHandle;
|
||||
|
||||
const DllHandle kInvalidDllHandle = NULL;
|
||||
#else
|
||||
#error Not implemented
|
||||
#endif
|
||||
|
||||
// These are helpers for use only by the class below.
|
||||
DllHandle InternalLoadDll(const char dll_name[]);
|
||||
|
||||
void InternalUnloadDll(DllHandle handle);
|
||||
|
||||
bool InternalLoadSymbols(DllHandle handle,
|
||||
int num_symbols,
|
||||
const char *const symbol_names[],
|
||||
void *symbols[]);
|
||||
|
||||
template <int SYMBOL_TABLE_SIZE,
|
||||
const char kDllName[],
|
||||
const char *const kSymbolNames[]>
|
||||
class LateBindingSymbolTable {
|
||||
public:
|
||||
LateBindingSymbolTable()
|
||||
: handle_(kInvalidDllHandle),
|
||||
undefined_symbols_(false) {
|
||||
memset(symbols_, 0, sizeof(symbols_));
|
||||
}
|
||||
|
||||
~LateBindingSymbolTable() {
|
||||
Unload();
|
||||
}
|
||||
|
||||
static int NumSymbols() {
|
||||
return SYMBOL_TABLE_SIZE;
|
||||
}
|
||||
|
||||
// We do not use this, but we offer it for theoretical convenience.
|
||||
static const char *GetSymbolName(int index) {
|
||||
assert(index < NumSymbols());
|
||||
return kSymbolNames[index];
|
||||
}
|
||||
|
||||
bool IsLoaded() const {
|
||||
return handle_ != kInvalidDllHandle;
|
||||
}
|
||||
|
||||
// Loads the DLL and the symbol table. Returns true iff the DLL and symbol
|
||||
// table loaded successfully.
|
||||
bool Load() {
|
||||
if (IsLoaded()) {
|
||||
return true;
|
||||
}
|
||||
if (undefined_symbols_) {
|
||||
// We do not attempt to load again because repeated attempts are not
|
||||
// likely to succeed and DLL loading is costly.
|
||||
//WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
|
||||
// "We know there are undefined symbols");
|
||||
return false;
|
||||
}
|
||||
handle_ = InternalLoadDll(kDllName);
|
||||
if (!IsLoaded()) {
|
||||
return false;
|
||||
}
|
||||
if (!InternalLoadSymbols(handle_, NumSymbols(), kSymbolNames, symbols_)) {
|
||||
undefined_symbols_ = true;
|
||||
Unload();
|
||||
return false;
|
||||
}
|
||||
return true;
|
||||
}
|
||||
|
||||
void Unload() {
|
||||
if (!IsLoaded()) {
|
||||
return;
|
||||
}
|
||||
InternalUnloadDll(handle_);
|
||||
handle_ = kInvalidDllHandle;
|
||||
memset(symbols_, 0, sizeof(symbols_));
|
||||
}
|
||||
|
||||
// Retrieves the given symbol. NOTE: Recommended to use LATESYM_GET below
|
||||
// instead of this.
|
||||
void *GetSymbol(int index) const {
|
||||
assert(IsLoaded());
|
||||
assert(index < NumSymbols());
|
||||
return symbols_[index];
|
||||
}
|
||||
|
||||
private:
|
||||
DllHandle handle_;
|
||||
bool undefined_symbols_;
|
||||
void *symbols_[SYMBOL_TABLE_SIZE];
|
||||
|
||||
DISALLOW_COPY_AND_ASSIGN(LateBindingSymbolTable);
|
||||
};
|
||||
|
||||
// This macro must be invoked in a header to declare a symbol table class.
|
||||
#define LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(ClassName) \
|
||||
enum {
|
||||
|
||||
// This macro must be invoked in the header declaration once for each symbol
|
||||
// (recommended to use an X-Macro to avoid duplication).
|
||||
// This macro defines an enum with names built from the symbols, which
|
||||
// essentially creates a hash table in the compiler from symbol names to their
|
||||
// indices in the symbol table class.
|
||||
#define LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(ClassName, sym) \
|
||||
ClassName##_SYMBOL_TABLE_INDEX_##sym,
|
||||
|
||||
// This macro completes the header declaration.
|
||||
#define LATE_BINDING_SYMBOL_TABLE_DECLARE_END(ClassName) \
|
||||
ClassName##_SYMBOL_TABLE_SIZE \
|
||||
}; \
|
||||
\
|
||||
extern const char ClassName##_kDllName[]; \
|
||||
extern const char *const \
|
||||
ClassName##_kSymbolNames[ClassName##_SYMBOL_TABLE_SIZE]; \
|
||||
\
|
||||
typedef ::webrtc_adm_linux::LateBindingSymbolTable<ClassName##_SYMBOL_TABLE_SIZE, \
|
||||
ClassName##_kDllName, \
|
||||
ClassName##_kSymbolNames> \
|
||||
ClassName;
|
||||
|
||||
// This macro must be invoked in a .cc file to define a previously-declared
|
||||
// symbol table class.
|
||||
#define LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(ClassName, dllName) \
|
||||
const char ClassName##_kDllName[] = dllName; \
|
||||
const char *const ClassName##_kSymbolNames[ClassName##_SYMBOL_TABLE_SIZE] = {
|
||||
|
||||
// This macro must be invoked in the .cc definition once for each symbol
|
||||
// (recommended to use an X-Macro to avoid duplication).
|
||||
// This would have to use the mangled name if we were to ever support C++
|
||||
// symbols.
|
||||
#define LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(ClassName, sym) \
|
||||
#sym,
|
||||
|
||||
#define LATE_BINDING_SYMBOL_TABLE_DEFINE_END(ClassName) \
|
||||
};
|
||||
|
||||
// Index of a given symbol in the given symbol table class.
|
||||
#define LATESYM_INDEXOF(ClassName, sym) \
|
||||
(ClassName##_SYMBOL_TABLE_INDEX_##sym)
|
||||
|
||||
// Returns a reference to the given late-binded symbol, with the correct type.
|
||||
#define LATESYM_GET(ClassName, inst, sym) \
|
||||
(*reinterpret_cast<typeof(&sym)>( \
|
||||
(inst)->GetSymbol(LATESYM_INDEXOF(ClassName, sym))))
|
||||
|
||||
} // namespace webrtc_adm_linux
|
||||
|
||||
#endif // WEBRTC_ADM_LATEBINDINGSYMBOLTABLE_H
|
||||
@ -1,39 +0,0 @@
|
||||
/*
|
||||
* libjingle
|
||||
* Copyright 2004--2010, Google Inc.
|
||||
*
|
||||
* Redistribution and use in source and binary forms, with or without
|
||||
* modification, are permitted provided that the following conditions are met:
|
||||
*
|
||||
* 1. Redistributions of source code must retain the above copyright notice,
|
||||
* this list of conditions and the following disclaimer.
|
||||
* 2. Redistributions in binary form must reproduce the above copyright notice,
|
||||
* this list of conditions and the following disclaimer in the documentation
|
||||
* and/or other materials provided with the distribution.
|
||||
* 3. The name of the author may not be used to endorse or promote products
|
||||
* derived from this software without specific prior written permission.
|
||||
*
|
||||
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
|
||||
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
|
||||
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
|
||||
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
|
||||
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
|
||||
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
|
||||
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
|
||||
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
|
||||
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
|
||||
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
||||
*/
|
||||
|
||||
#include "pulseaudiosymboltable.h"
|
||||
|
||||
namespace webrtc_adm_linux_pulse {
|
||||
|
||||
LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(PulseAudioSymbolTable, "libpulse.so.0")
|
||||
#define X(sym) \
|
||||
LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(PulseAudioSymbolTable, sym)
|
||||
PULSE_AUDIO_SYMBOLS_LIST
|
||||
#undef X
|
||||
LATE_BINDING_SYMBOL_TABLE_DEFINE_END(PulseAudioSymbolTable)
|
||||
|
||||
} // namespace webrtc_adm_linux_pulse
|
||||
@ -1,104 +0,0 @@
|
||||
/*
|
||||
* libjingle
|
||||
* Copyright 2004--2010, Google Inc.
|
||||
*
|
||||
* Redistribution and use in source and binary forms, with or without
|
||||
* modification, are permitted provided that the following conditions are met:
|
||||
*
|
||||
* 1. Redistributions of source code must retain the above copyright notice,
|
||||
* this list of conditions and the following disclaimer.
|
||||
* 2. Redistributions in binary form must reproduce the above copyright notice,
|
||||
* this list of conditions and the following disclaimer in the documentation
|
||||
* and/or other materials provided with the distribution.
|
||||
* 3. The name of the author may not be used to endorse or promote products
|
||||
* derived from this software without specific prior written permission.
|
||||
*
|
||||
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
|
||||
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
|
||||
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
|
||||
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
|
||||
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
|
||||
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
|
||||
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
|
||||
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
|
||||
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
|
||||
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
||||
*/
|
||||
|
||||
#ifndef WEBRTC_AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_H
|
||||
#define WEBRTC_AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_H
|
||||
|
||||
#include "latebindingsymboltable.h"
|
||||
|
||||
namespace webrtc_adm_linux_pulse {
|
||||
|
||||
// The PulseAudio symbols we need, as an X-Macro list.
|
||||
// This list must contain precisely every libpulse function that is used in
|
||||
// the ADM LINUX PULSE Device and Mixer classes
|
||||
#define PULSE_AUDIO_SYMBOLS_LIST \
|
||||
X(pa_bytes_per_second) \
|
||||
X(pa_context_connect) \
|
||||
X(pa_context_disconnect) \
|
||||
X(pa_context_errno) \
|
||||
X(pa_context_get_protocol_version) \
|
||||
X(pa_context_get_server_info) \
|
||||
X(pa_context_get_sink_info_list) \
|
||||
X(pa_context_get_sink_info_by_index) \
|
||||
X(pa_context_get_sink_info_by_name) \
|
||||
X(pa_context_get_sink_input_info) \
|
||||
X(pa_context_get_source_info_by_index) \
|
||||
X(pa_context_get_source_info_by_name) \
|
||||
X(pa_context_get_source_info_list) \
|
||||
X(pa_context_get_state) \
|
||||
X(pa_context_new) \
|
||||
X(pa_context_set_sink_input_volume) \
|
||||
X(pa_context_set_sink_input_mute) \
|
||||
X(pa_context_set_source_volume_by_index) \
|
||||
X(pa_context_set_source_mute_by_index) \
|
||||
X(pa_context_set_state_callback) \
|
||||
X(pa_context_unref) \
|
||||
X(pa_cvolume_set) \
|
||||
X(pa_operation_get_state) \
|
||||
X(pa_operation_unref) \
|
||||
X(pa_stream_connect_playback) \
|
||||
X(pa_stream_connect_record) \
|
||||
X(pa_stream_disconnect) \
|
||||
X(pa_stream_drop) \
|
||||
X(pa_stream_get_device_index) \
|
||||
X(pa_stream_get_index) \
|
||||
X(pa_stream_get_latency) \
|
||||
X(pa_stream_get_sample_spec) \
|
||||
X(pa_stream_get_state) \
|
||||
X(pa_stream_new) \
|
||||
X(pa_stream_peek) \
|
||||
X(pa_stream_readable_size) \
|
||||
X(pa_stream_set_buffer_attr) \
|
||||
X(pa_stream_set_overflow_callback) \
|
||||
X(pa_stream_set_read_callback) \
|
||||
X(pa_stream_set_state_callback) \
|
||||
X(pa_stream_set_underflow_callback) \
|
||||
X(pa_stream_set_write_callback) \
|
||||
X(pa_stream_unref) \
|
||||
X(pa_stream_writable_size) \
|
||||
X(pa_stream_write) \
|
||||
X(pa_strerror) \
|
||||
X(pa_threaded_mainloop_free) \
|
||||
X(pa_threaded_mainloop_get_api) \
|
||||
X(pa_threaded_mainloop_lock) \
|
||||
X(pa_threaded_mainloop_new) \
|
||||
X(pa_threaded_mainloop_signal) \
|
||||
X(pa_threaded_mainloop_start) \
|
||||
X(pa_threaded_mainloop_stop) \
|
||||
X(pa_threaded_mainloop_unlock) \
|
||||
X(pa_threaded_mainloop_wait)
|
||||
|
||||
LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(PulseAudioSymbolTable)
|
||||
#define X(sym) \
|
||||
LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(PulseAudioSymbolTable, sym)
|
||||
PULSE_AUDIO_SYMBOLS_LIST
|
||||
#undef X
|
||||
LATE_BINDING_SYMBOL_TABLE_DECLARE_END(PulseAudioSymbolTable)
|
||||
|
||||
} // namespace webrtc_adm_linux_pulse
|
||||
|
||||
#endif // WEBRTC_AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_H
|
||||
File diff suppressed because it is too large
Load Diff
@ -1,400 +0,0 @@
|
||||
/*
|
||||
* 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.
|
||||
*/
|
||||
|
||||
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_MAC_H
|
||||
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_MAC_H
|
||||
|
||||
#include "audio_device_generic.h"
|
||||
#include "critical_section_wrapper.h"
|
||||
#include "audio_mixer_manager_mac.h"
|
||||
|
||||
#include <CoreAudio/CoreAudio.h>
|
||||
#include <AudioToolbox/AudioConverter.h>
|
||||
#include <mach/semaphore.h>
|
||||
|
||||
struct PaUtilRingBuffer;
|
||||
|
||||
namespace webrtc
|
||||
{
|
||||
class EventWrapper;
|
||||
class ThreadWrapper;
|
||||
|
||||
const WebRtc_UWord32 N_REC_SAMPLES_PER_SEC = 48000;
|
||||
const WebRtc_UWord32 N_PLAY_SAMPLES_PER_SEC = 48000;
|
||||
|
||||
const WebRtc_UWord32 N_REC_CHANNELS = 1; // default is mono recording
|
||||
const WebRtc_UWord32 N_PLAY_CHANNELS = 2; // default is stereo playout
|
||||
const WebRtc_UWord32 N_DEVICE_CHANNELS = 8;
|
||||
|
||||
const WebRtc_UWord32 ENGINE_REC_BUF_SIZE_IN_SAMPLES = (N_REC_SAMPLES_PER_SEC
|
||||
/ 100);
|
||||
const WebRtc_UWord32 ENGINE_PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC
|
||||
/ 100);
|
||||
|
||||
enum
|
||||
{
|
||||
N_BLOCKS_IO = 2
|
||||
};
|
||||
enum
|
||||
{
|
||||
N_BUFFERS_IN = 10
|
||||
};
|
||||
enum
|
||||
{
|
||||
N_BUFFERS_OUT = 3
|
||||
}; // Must be at least N_BLOCKS_IO
|
||||
|
||||
const WebRtc_UWord32 TIMER_PERIOD_MS = (2 * 10 * N_BLOCKS_IO * 1000000);
|
||||
|
||||
const WebRtc_UWord32 REC_BUF_SIZE_IN_SAMPLES = (ENGINE_REC_BUF_SIZE_IN_SAMPLES
|
||||
* N_DEVICE_CHANNELS * N_BUFFERS_IN);
|
||||
const WebRtc_UWord32 PLAY_BUF_SIZE_IN_SAMPLES =
|
||||
(ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * N_PLAY_CHANNELS * N_BUFFERS_OUT);
|
||||
|
||||
class AudioDeviceMac: public AudioDeviceGeneric
|
||||
{
|
||||
public:
|
||||
AudioDeviceMac(const WebRtc_Word32 id);
|
||||
~AudioDeviceMac();
|
||||
|
||||
// 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;
|
||||
|
||||
// Microphone mute control
|
||||
virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
|
||||
virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
|
||||
virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
|
||||
|
||||
// Speaker mute control
|
||||
virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
|
||||
virtual WebRtc_Word32 SetSpeakerMute(bool enable);
|
||||
virtual WebRtc_Word32 SpeakerMute(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;
|
||||
|
||||
public:
|
||||
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();
|
||||
|
||||
public:
|
||||
virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
|
||||
|
||||
private:
|
||||
void Lock()
|
||||
{
|
||||
_critSect.Enter();
|
||||
}
|
||||
;
|
||||
void UnLock()
|
||||
{
|
||||
_critSect.Leave();
|
||||
}
|
||||
;
|
||||
WebRtc_Word32 Id()
|
||||
{
|
||||
return _id;
|
||||
}
|
||||
|
||||
static void AtomicSet32(int32_t* theValue, int32_t newValue);
|
||||
static int32_t AtomicGet32(int32_t* theValue);
|
||||
|
||||
static void logCAMsg(const TraceLevel level,
|
||||
const TraceModule module,
|
||||
const WebRtc_Word32 id, const char *msg,
|
||||
const char *err);
|
||||
|
||||
WebRtc_Word32 GetNumberDevices(const AudioObjectPropertyScope scope,
|
||||
AudioDeviceID scopedDeviceIds[],
|
||||
const WebRtc_UWord32 deviceListLength);
|
||||
|
||||
WebRtc_Word32 GetDeviceName(const AudioObjectPropertyScope scope,
|
||||
const WebRtc_UWord16 index, char* name);
|
||||
|
||||
WebRtc_Word32 InitDevice(WebRtc_UWord16 userDeviceIndex,
|
||||
AudioDeviceID& deviceId, bool isInput);
|
||||
|
||||
static OSStatus
|
||||
objectListenerProc(AudioObjectID objectId, UInt32 numberAddresses,
|
||||
const AudioObjectPropertyAddress addresses[],
|
||||
void* clientData);
|
||||
|
||||
OSStatus
|
||||
implObjectListenerProc(AudioObjectID objectId, UInt32 numberAddresses,
|
||||
const AudioObjectPropertyAddress addresses[]);
|
||||
|
||||
WebRtc_Word32 HandleDeviceChange();
|
||||
|
||||
WebRtc_Word32
|
||||
HandleStreamFormatChange(AudioObjectID objectId,
|
||||
AudioObjectPropertyAddress propertyAddress);
|
||||
|
||||
WebRtc_Word32
|
||||
HandleDataSourceChange(AudioObjectID objectId,
|
||||
AudioObjectPropertyAddress propertyAddress);
|
||||
|
||||
WebRtc_Word32
|
||||
HandleProcessorOverload(AudioObjectPropertyAddress propertyAddress);
|
||||
|
||||
private:
|
||||
static OSStatus deviceIOProc(AudioDeviceID device,
|
||||
const AudioTimeStamp *now,
|
||||
const AudioBufferList *inputData,
|
||||
const AudioTimeStamp *inputTime,
|
||||
AudioBufferList *outputData,
|
||||
const AudioTimeStamp* outputTime,
|
||||
void *clientData);
|
||||
|
||||
static OSStatus
|
||||
outConverterProc(AudioConverterRef audioConverter,
|
||||
UInt32 *numberDataPackets, AudioBufferList *data,
|
||||
AudioStreamPacketDescription **dataPacketDescription,
|
||||
void *userData);
|
||||
|
||||
static OSStatus inDeviceIOProc(AudioDeviceID device,
|
||||
const AudioTimeStamp *now,
|
||||
const AudioBufferList *inputData,
|
||||
const AudioTimeStamp *inputTime,
|
||||
AudioBufferList *outputData,
|
||||
const AudioTimeStamp *outputTime,
|
||||
void *clientData);
|
||||
|
||||
static OSStatus
|
||||
inConverterProc(AudioConverterRef audioConverter,
|
||||
UInt32 *numberDataPackets, AudioBufferList *data,
|
||||
AudioStreamPacketDescription **dataPacketDescription,
|
||||
void *inUserData);
|
||||
|
||||
OSStatus implDeviceIOProc(const AudioBufferList *inputData,
|
||||
const AudioTimeStamp *inputTime,
|
||||
AudioBufferList *outputData,
|
||||
const AudioTimeStamp *outputTime);
|
||||
|
||||
OSStatus implOutConverterProc(UInt32 *numberDataPackets,
|
||||
AudioBufferList *data);
|
||||
|
||||
OSStatus implInDeviceIOProc(const AudioBufferList *inputData,
|
||||
const AudioTimeStamp *inputTime);
|
||||
|
||||
OSStatus implInConverterProc(UInt32 *numberDataPackets,
|
||||
AudioBufferList *data);
|
||||
|
||||
static bool RunCapture(void*);
|
||||
static bool RunRender(void*);
|
||||
bool CaptureWorkerThread();
|
||||
bool RenderWorkerThread();
|
||||
|
||||
private:
|
||||
AudioDeviceBuffer* _ptrAudioBuffer;
|
||||
|
||||
CriticalSectionWrapper& _critSect;
|
||||
CriticalSectionWrapper& _critSectCb;
|
||||
|
||||
EventWrapper& _stopEventRec;
|
||||
EventWrapper& _stopEvent;
|
||||
|
||||
ThreadWrapper* _captureWorkerThread;
|
||||
ThreadWrapper* _renderWorkerThread;
|
||||
WebRtc_UWord32 _captureWorkerThreadId;
|
||||
WebRtc_UWord32 _renderWorkerThreadId;
|
||||
|
||||
WebRtc_Word32 _id;
|
||||
|
||||
AudioMixerManagerMac _mixerManager;
|
||||
|
||||
WebRtc_UWord16 _inputDeviceIndex;
|
||||
WebRtc_UWord16 _outputDeviceIndex;
|
||||
AudioDeviceID _inputDeviceID;
|
||||
AudioDeviceID _outputDeviceID;
|
||||
#if __MAC_OS_X_VERSION_MAX_ALLOWED >= 1050
|
||||
AudioDeviceIOProcID _inDeviceIOProcID;
|
||||
AudioDeviceIOProcID _deviceIOProcID;
|
||||
#endif
|
||||
bool _inputDeviceIsSpecified;
|
||||
bool _outputDeviceIsSpecified;
|
||||
|
||||
WebRtc_UWord8 _recChannels;
|
||||
WebRtc_UWord8 _playChannels;
|
||||
|
||||
Float32* _captureBufData;
|
||||
SInt16* _renderBufData;
|
||||
|
||||
SInt16 _renderConvertData[PLAY_BUF_SIZE_IN_SAMPLES];
|
||||
|
||||
AudioDeviceModule::BufferType _playBufType;
|
||||
|
||||
private:
|
||||
bool _initialized;
|
||||
bool _isShutDown;
|
||||
bool _recording;
|
||||
bool _playing;
|
||||
bool _recIsInitialized;
|
||||
bool _playIsInitialized;
|
||||
bool _startRec;
|
||||
bool _stopRec;
|
||||
bool _stopPlay;
|
||||
bool _AGC;
|
||||
|
||||
// Atomically set varaibles
|
||||
int32_t _renderDeviceIsAlive;
|
||||
int32_t _captureDeviceIsAlive;
|
||||
|
||||
bool _twoDevices;
|
||||
bool _doStop; // For play if not shared device or play+rec if shared device
|
||||
bool _doStopRec; // For rec if not shared device
|
||||
bool _macBookPro;
|
||||
bool _macBookProPanRight;
|
||||
bool _stereoRender;
|
||||
bool _stereoRenderRequested;
|
||||
|
||||
AudioConverterRef _captureConverter;
|
||||
AudioConverterRef _renderConverter;
|
||||
|
||||
AudioStreamBasicDescription _outStreamFormat;
|
||||
AudioStreamBasicDescription _outDesiredFormat;
|
||||
AudioStreamBasicDescription _inStreamFormat;
|
||||
AudioStreamBasicDescription _inDesiredFormat;
|
||||
|
||||
WebRtc_UWord32 _captureLatencyUs;
|
||||
WebRtc_UWord32 _renderLatencyUs;
|
||||
|
||||
// Atomically set variables
|
||||
mutable int32_t _captureDelayUs;
|
||||
mutable int32_t _renderDelayUs;
|
||||
|
||||
WebRtc_Word32 _renderDelayOffsetSamples;
|
||||
|
||||
private:
|
||||
WebRtc_UWord16 _playBufDelay; // playback delay
|
||||
WebRtc_UWord16 _playBufDelayFixed; // fixed playback delay
|
||||
|
||||
WebRtc_UWord16 _playWarning;
|
||||
WebRtc_UWord16 _playError;
|
||||
WebRtc_UWord16 _recWarning;
|
||||
WebRtc_UWord16 _recError;
|
||||
|
||||
PaUtilRingBuffer* _paCaptureBuffer;
|
||||
PaUtilRingBuffer* _paRenderBuffer;
|
||||
|
||||
semaphore_t _renderSemaphore;
|
||||
semaphore_t _captureSemaphore;
|
||||
|
||||
WebRtc_UWord32 _captureBufSizeSamples;
|
||||
WebRtc_UWord32 _renderBufSizeSamples;
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_MAC_H_
|
||||
@ -1,56 +0,0 @@
|
||||
/*
|
||||
* 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_utility_mac.h"
|
||||
#include "audio_device_config.h" // DEBUG_PRINT()
|
||||
#include "critical_section_wrapper.h"
|
||||
#include "trace.h"
|
||||
|
||||
namespace webrtc
|
||||
{
|
||||
|
||||
AudioDeviceUtilityMac::AudioDeviceUtilityMac(const WebRtc_Word32 id) :
|
||||
_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
|
||||
_id(id),
|
||||
_lastError(AudioDeviceModule::kAdmErrNone)
|
||||
{
|
||||
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
|
||||
"%s created", __FUNCTION__);
|
||||
}
|
||||
|
||||
// ----------------------------------------------------------------------------
|
||||
// AudioDeviceUtilityMac() - dtor
|
||||
// ----------------------------------------------------------------------------
|
||||
|
||||
AudioDeviceUtilityMac::~AudioDeviceUtilityMac()
|
||||
{
|
||||
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
|
||||
"%s destroyed", __FUNCTION__);
|
||||
{
|
||||
CriticalSectionScoped lock(_critSect);
|
||||
|
||||
// free stuff here...
|
||||
}
|
||||
|
||||
delete &_critSect;
|
||||
}
|
||||
|
||||
WebRtc_Word32 AudioDeviceUtilityMac::Init()
|
||||
{
|
||||
WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
|
||||
"%s", __FUNCTION__);
|
||||
|
||||
WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
|
||||
" OS info: %s", "OS X");
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
@ -1,37 +0,0 @@
|
||||
/*
|
||||
* 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.
|
||||
*/
|
||||
|
||||
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_MAC_H
|
||||
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_MAC_H
|
||||
|
||||
#include "audio_device_utility.h"
|
||||
#include "audio_device.h"
|
||||
|
||||
namespace webrtc
|
||||
{
|
||||
class CriticalSectionWrapper;
|
||||
|
||||
class AudioDeviceUtilityMac: public AudioDeviceUtility
|
||||
{
|
||||
public:
|
||||
AudioDeviceUtilityMac(const WebRtc_Word32 id);
|
||||
~AudioDeviceUtilityMac();
|
||||
|
||||
virtual WebRtc_Word32 Init();
|
||||
|
||||
private:
|
||||
CriticalSectionWrapper& _critSect;
|
||||
WebRtc_Word32 _id;
|
||||
AudioDeviceModule::ErrorCode _lastError;
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_UTILITY_MAC_H_
|
||||
File diff suppressed because it is too large
Load Diff
@ -1,80 +0,0 @@
|
||||
/*
|
||||
* 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.
|
||||
*/
|
||||
|
||||
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_MAC_H
|
||||
#define WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_MAC_H
|
||||
|
||||
#include "typedefs.h"
|
||||
#include "audio_device.h"
|
||||
#include "critical_section_wrapper.h"
|
||||
|
||||
#include <CoreAudio/CoreAudio.h>
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
class AudioMixerManagerMac
|
||||
{
|
||||
public:
|
||||
WebRtc_Word32 OpenSpeaker(AudioDeviceID deviceID);
|
||||
WebRtc_Word32 OpenMicrophone(AudioDeviceID deviceID);
|
||||
WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
|
||||
WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
|
||||
WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
|
||||
WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
|
||||
WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
|
||||
WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
|
||||
WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
|
||||
WebRtc_Word32 SetSpeakerMute(bool enable);
|
||||
WebRtc_Word32 SpeakerMute(bool& enabled) const;
|
||||
WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
|
||||
WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
|
||||
WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
|
||||
WebRtc_Word32 SetMicrophoneMute(bool enable);
|
||||
WebRtc_Word32 MicrophoneMute(bool& enabled) const;
|
||||
WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
|
||||
WebRtc_Word32 SetMicrophoneBoost(bool enable);
|
||||
WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
|
||||
WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
|
||||
WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
|
||||
WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
|
||||
WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
|
||||
WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
|
||||
WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
|
||||
WebRtc_Word32 Close();
|
||||
WebRtc_Word32 CloseSpeaker();
|
||||
WebRtc_Word32 CloseMicrophone();
|
||||
bool SpeakerIsInitialized() const;
|
||||
bool MicrophoneIsInitialized() const;
|
||||
|
||||
public:
|
||||
AudioMixerManagerMac(const WebRtc_Word32 id);
|
||||
~AudioMixerManagerMac();
|
||||
|
||||
private:
|
||||
static void logCAMsg(const TraceLevel level,
|
||||
const TraceModule module,
|
||||
const WebRtc_Word32 id, const char *msg,
|
||||
const char *err);
|
||||
|
||||
private:
|
||||
CriticalSectionWrapper& _critSect;
|
||||
WebRtc_Word32 _id;
|
||||
|
||||
AudioDeviceID _inputDeviceID;
|
||||
AudioDeviceID _outputDeviceID;
|
||||
|
||||
WebRtc_UWord16 _noInputChannels;
|
||||
WebRtc_UWord16 _noOutputChannels;
|
||||
|
||||
};
|
||||
|
||||
} //namespace webrtc
|
||||
|
||||
#endif // AUDIO_MIXER_MAC_H
|
||||
@ -1,127 +0,0 @@
|
||||
/*
|
||||
* $Id: pa_memorybarrier.h 1240 2007-07-17 13:05:07Z bjornroche $
|
||||
* Portable Audio I/O Library
|
||||
* Memory barrier utilities
|
||||
*
|
||||
* Author: Bjorn Roche, XO Audio, LLC
|
||||
*
|
||||
* This program uses the PortAudio Portable Audio Library.
|
||||
* For more information see: http://www.portaudio.com
|
||||
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
|
||||
*
|
||||
* Permission is hereby granted, free of charge, to any person obtaining
|
||||
* a copy of this software and associated documentation files
|
||||
* (the "Software"), to deal in the Software without restriction,
|
||||
* including without limitation the rights to use, copy, modify, merge,
|
||||
* publish, distribute, sublicense, and/or sell copies of the Software,
|
||||
* and to permit persons to whom the Software is furnished to do so,
|
||||
* subject to the following conditions:
|
||||
*
|
||||
* The above copyright notice and this permission notice shall be
|
||||
* included in all copies or substantial portions of the Software.
|
||||
*
|
||||
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
|
||||
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
|
||||
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
|
||||
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
|
||||
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
|
||||
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
|
||||
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
|
||||
*/
|
||||
|
||||
/*
|
||||
* The text above constitutes the entire PortAudio license; however,
|
||||
* the PortAudio community also makes the following non-binding requests:
|
||||
*
|
||||
* Any person wishing to distribute modifications to the Software is
|
||||
* requested to send the modifications to the original developer so that
|
||||
* they can be incorporated into the canonical version. It is also
|
||||
* requested that these non-binding requests be included along with the
|
||||
* license above.
|
||||
*/
|
||||
|
||||
/**
|
||||
@file pa_memorybarrier.h
|
||||
@ingroup common_src
|
||||
*/
|
||||
|
||||
/****************
|
||||
* Some memory barrier primitives based on the system.
|
||||
* right now only OS X, FreeBSD, and Linux are supported. In addition to providing
|
||||
* memory barriers, these functions should ensure that data cached in registers
|
||||
* is written out to cache where it can be snooped by other CPUs. (ie, the volatile
|
||||
* keyword should not be required)
|
||||
*
|
||||
* the primitives that must be defined are:
|
||||
*
|
||||
* PaUtil_FullMemoryBarrier()
|
||||
* PaUtil_ReadMemoryBarrier()
|
||||
* PaUtil_WriteMemoryBarrier()
|
||||
*
|
||||
****************/
|
||||
|
||||
#if defined(__APPLE__)
|
||||
# include <libkern/OSAtomic.h>
|
||||
/* Here are the memory barrier functions. Mac OS X only provides
|
||||
full memory barriers, so the three types of barriers are the same,
|
||||
however, these barriers are superior to compiler-based ones. */
|
||||
# define PaUtil_FullMemoryBarrier() OSMemoryBarrier()
|
||||
# define PaUtil_ReadMemoryBarrier() OSMemoryBarrier()
|
||||
# define PaUtil_WriteMemoryBarrier() OSMemoryBarrier()
|
||||
#elif defined(__GNUC__)
|
||||
/* GCC >= 4.1 has built-in intrinsics. We'll use those */
|
||||
# if (__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 1)
|
||||
# define PaUtil_FullMemoryBarrier() __sync_synchronize()
|
||||
# define PaUtil_ReadMemoryBarrier() __sync_synchronize()
|
||||
# define PaUtil_WriteMemoryBarrier() __sync_synchronize()
|
||||
/* as a fallback, GCC understands volatile asm and "memory" to mean it
|
||||
* should not reorder memory read/writes */
|
||||
/* Note that it is not clear that any compiler actually defines __PPC__,
|
||||
* it can probably removed safely. */
|
||||
# elif defined( __ppc__ ) || defined( __powerpc__) || defined( __PPC__ )
|
||||
# define PaUtil_FullMemoryBarrier() asm volatile("sync":::"memory")
|
||||
# define PaUtil_ReadMemoryBarrier() asm volatile("sync":::"memory")
|
||||
# define PaUtil_WriteMemoryBarrier() asm volatile("sync":::"memory")
|
||||
# elif defined( __i386__ ) || defined( __i486__ ) || defined( __i586__ ) || \
|
||||
defined( __i686__ ) || defined( __x86_64__ )
|
||||
# define PaUtil_FullMemoryBarrier() asm volatile("mfence":::"memory")
|
||||
# define PaUtil_ReadMemoryBarrier() asm volatile("lfence":::"memory")
|
||||
# define PaUtil_WriteMemoryBarrier() asm volatile("sfence":::"memory")
|
||||
# else
|
||||
# ifdef ALLOW_SMP_DANGERS
|
||||
# warning Memory barriers not defined on this system or system unknown
|
||||
# warning For SMP safety, you should fix this.
|
||||
# define PaUtil_FullMemoryBarrier()
|
||||
# define PaUtil_ReadMemoryBarrier()
|
||||
# define PaUtil_WriteMemoryBarrier()
|
||||
# else
|
||||
# error Memory barriers are not defined on this system. You can still compile by defining ALLOW_SMP_DANGERS, but SMP safety will not be guaranteed.
|
||||
# endif
|
||||
# endif
|
||||
#elif (_MSC_VER >= 1400) && !defined(_WIN32_WCE)
|
||||
# include <intrin.h>
|
||||
# pragma intrinsic(_ReadWriteBarrier)
|
||||
# pragma intrinsic(_ReadBarrier)
|
||||
# pragma intrinsic(_WriteBarrier)
|
||||
# define PaUtil_FullMemoryBarrier() _ReadWriteBarrier()
|
||||
# define PaUtil_ReadMemoryBarrier() _ReadBarrier()
|
||||
# define PaUtil_WriteMemoryBarrier() _WriteBarrier()
|
||||
#elif defined(_WIN32_WCE)
|
||||
# define PaUtil_FullMemoryBarrier()
|
||||
# define PaUtil_ReadMemoryBarrier()
|
||||
# define PaUtil_WriteMemoryBarrier()
|
||||
#elif defined(_MSC_VER) || defined(__BORLANDC__)
|
||||
# define PaUtil_FullMemoryBarrier() _asm { lock add [esp], 0 }
|
||||
# define PaUtil_ReadMemoryBarrier() _asm { lock add [esp], 0 }
|
||||
# define PaUtil_WriteMemoryBarrier() _asm { lock add [esp], 0 }
|
||||
#else
|
||||
# ifdef ALLOW_SMP_DANGERS
|
||||
# warning Memory barriers not defined on this system or system unknown
|
||||
# warning For SMP safety, you should fix this.
|
||||
# define PaUtil_FullMemoryBarrier()
|
||||
# define PaUtil_ReadMemoryBarrier()
|
||||
# define PaUtil_WriteMemoryBarrier()
|
||||
# else
|
||||
# error Memory barriers are not defined on this system. You can still compile by defining ALLOW_SMP_DANGERS, but SMP safety will not be guaranteed.
|
||||
# endif
|
||||
#endif
|
||||
@ -1,227 +0,0 @@
|
||||
/*
|
||||
* $Id: pa_ringbuffer.c 1421 2009-11-18 16:09:05Z bjornroche $
|
||||
* Portable Audio I/O Library
|
||||
* Ring Buffer utility.
|
||||
*
|
||||
* Author: Phil Burk, http://www.softsynth.com
|
||||
* modified for SMP safety on Mac OS X by Bjorn Roche
|
||||
* modified for SMP safety on Linux by Leland Lucius
|
||||
* also, allowed for const where possible
|
||||
* modified for multiple-byte-sized data elements by Sven Fischer
|
||||
*
|
||||
* Note that this is safe only for a single-thread reader and a
|
||||
* single-thread writer.
|
||||
*
|
||||
* This program uses the PortAudio Portable Audio Library.
|
||||
* For more information see: http://www.portaudio.com
|
||||
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
|
||||
*
|
||||
* Permission is hereby granted, free of charge, to any person obtaining
|
||||
* a copy of this software and associated documentation files
|
||||
* (the "Software"), to deal in the Software without restriction,
|
||||
* including without limitation the rights to use, copy, modify, merge,
|
||||
* publish, distribute, sublicense, and/or sell copies of the Software,
|
||||
* and to permit persons to whom the Software is furnished to do so,
|
||||
* subject to the following conditions:
|
||||
*
|
||||
* The above copyright notice and this permission notice shall be
|
||||
* included in all copies or substantial portions of the Software.
|
||||
*
|
||||
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
|
||||
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
|
||||
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
|
||||
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
|
||||
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
|
||||
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
|
||||
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
|
||||
*/
|
||||
|
||||
/*
|
||||
* The text above constitutes the entire PortAudio license; however,
|
||||
* the PortAudio community also makes the following non-binding requests:
|
||||
*
|
||||
* Any person wishing to distribute modifications to the Software is
|
||||
* requested to send the modifications to the original developer so that
|
||||
* they can be incorporated into the canonical version. It is also
|
||||
* requested that these non-binding requests be included along with the
|
||||
* license above.
|
||||
*/
|
||||
|
||||
/**
|
||||
@file
|
||||
@ingroup common_src
|
||||
*/
|
||||
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
#include <math.h>
|
||||
#include "pa_ringbuffer.h"
|
||||
#include <string.h>
|
||||
#include "pa_memorybarrier.h"
|
||||
|
||||
/***************************************************************************
|
||||
* Initialize FIFO.
|
||||
* elementCount must be power of 2, returns -1 if not.
|
||||
*/
|
||||
ring_buffer_size_t PaUtil_InitializeRingBuffer( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementSizeBytes, ring_buffer_size_t elementCount, void *dataPtr )
|
||||
{
|
||||
if( ((elementCount-1) & elementCount) != 0) return -1; /* Not Power of two. */
|
||||
rbuf->bufferSize = elementCount;
|
||||
rbuf->buffer = (char *)dataPtr;
|
||||
PaUtil_FlushRingBuffer( rbuf );
|
||||
rbuf->bigMask = (elementCount*2)-1;
|
||||
rbuf->smallMask = (elementCount)-1;
|
||||
rbuf->elementSizeBytes = elementSizeBytes;
|
||||
return 0;
|
||||
}
|
||||
|
||||
/***************************************************************************
|
||||
** Return number of elements available for reading. */
|
||||
ring_buffer_size_t PaUtil_GetRingBufferReadAvailable( PaUtilRingBuffer *rbuf )
|
||||
{
|
||||
PaUtil_ReadMemoryBarrier();
|
||||
return ( (rbuf->writeIndex - rbuf->readIndex) & rbuf->bigMask );
|
||||
}
|
||||
/***************************************************************************
|
||||
** Return number of elements available for writing. */
|
||||
ring_buffer_size_t PaUtil_GetRingBufferWriteAvailable( PaUtilRingBuffer *rbuf )
|
||||
{
|
||||
/* Since we are calling PaUtil_GetRingBufferReadAvailable, we don't need an aditional MB */
|
||||
return ( rbuf->bufferSize - PaUtil_GetRingBufferReadAvailable(rbuf));
|
||||
}
|
||||
|
||||
/***************************************************************************
|
||||
** Clear buffer. Should only be called when buffer is NOT being read. */
|
||||
void PaUtil_FlushRingBuffer( PaUtilRingBuffer *rbuf )
|
||||
{
|
||||
rbuf->writeIndex = rbuf->readIndex = 0;
|
||||
}
|
||||
|
||||
/***************************************************************************
|
||||
** Get address of region(s) to which we can write data.
|
||||
** If the region is contiguous, size2 will be zero.
|
||||
** If non-contiguous, size2 will be the size of second region.
|
||||
** Returns room available to be written or elementCount, whichever is smaller.
|
||||
*/
|
||||
ring_buffer_size_t PaUtil_GetRingBufferWriteRegions( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount,
|
||||
void **dataPtr1, ring_buffer_size_t *sizePtr1,
|
||||
void **dataPtr2, ring_buffer_size_t *sizePtr2 )
|
||||
{
|
||||
ring_buffer_size_t index;
|
||||
ring_buffer_size_t available = PaUtil_GetRingBufferWriteAvailable( rbuf );
|
||||
if( elementCount > available ) elementCount = available;
|
||||
/* Check to see if write is not contiguous. */
|
||||
index = rbuf->writeIndex & rbuf->smallMask;
|
||||
if( (index + elementCount) > rbuf->bufferSize )
|
||||
{
|
||||
/* Write data in two blocks that wrap the buffer. */
|
||||
ring_buffer_size_t firstHalf = rbuf->bufferSize - index;
|
||||
*dataPtr1 = &rbuf->buffer[index*rbuf->elementSizeBytes];
|
||||
*sizePtr1 = firstHalf;
|
||||
*dataPtr2 = &rbuf->buffer[0];
|
||||
*sizePtr2 = elementCount - firstHalf;
|
||||
}
|
||||
else
|
||||
{
|
||||
*dataPtr1 = &rbuf->buffer[index*rbuf->elementSizeBytes];
|
||||
*sizePtr1 = elementCount;
|
||||
*dataPtr2 = NULL;
|
||||
*sizePtr2 = 0;
|
||||
}
|
||||
return elementCount;
|
||||
}
|
||||
|
||||
|
||||
/***************************************************************************
|
||||
*/
|
||||
ring_buffer_size_t PaUtil_AdvanceRingBufferWriteIndex( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount )
|
||||
{
|
||||
/* we need to ensure that previous writes are seen before we update the write index */
|
||||
PaUtil_WriteMemoryBarrier();
|
||||
return rbuf->writeIndex = (rbuf->writeIndex + elementCount) & rbuf->bigMask;
|
||||
}
|
||||
|
||||
/***************************************************************************
|
||||
** Get address of region(s) from which we can read data.
|
||||
** If the region is contiguous, size2 will be zero.
|
||||
** If non-contiguous, size2 will be the size of second region.
|
||||
** Returns room available to be written or elementCount, whichever is smaller.
|
||||
*/
|
||||
ring_buffer_size_t PaUtil_GetRingBufferReadRegions( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount,
|
||||
void **dataPtr1, ring_buffer_size_t *sizePtr1,
|
||||
void **dataPtr2, ring_buffer_size_t *sizePtr2 )
|
||||
{
|
||||
ring_buffer_size_t index;
|
||||
ring_buffer_size_t available = PaUtil_GetRingBufferReadAvailable( rbuf );
|
||||
if( elementCount > available ) elementCount = available;
|
||||
/* Check to see if read is not contiguous. */
|
||||
index = rbuf->readIndex & rbuf->smallMask;
|
||||
if( (index + elementCount) > rbuf->bufferSize )
|
||||
{
|
||||
/* Write data in two blocks that wrap the buffer. */
|
||||
ring_buffer_size_t firstHalf = rbuf->bufferSize - index;
|
||||
*dataPtr1 = &rbuf->buffer[index*rbuf->elementSizeBytes];
|
||||
*sizePtr1 = firstHalf;
|
||||
*dataPtr2 = &rbuf->buffer[0];
|
||||
*sizePtr2 = elementCount - firstHalf;
|
||||
}
|
||||
else
|
||||
{
|
||||
*dataPtr1 = &rbuf->buffer[index*rbuf->elementSizeBytes];
|
||||
*sizePtr1 = elementCount;
|
||||
*dataPtr2 = NULL;
|
||||
*sizePtr2 = 0;
|
||||
}
|
||||
return elementCount;
|
||||
}
|
||||
/***************************************************************************
|
||||
*/
|
||||
ring_buffer_size_t PaUtil_AdvanceRingBufferReadIndex( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount )
|
||||
{
|
||||
/* we need to ensure that previous writes are always seen before updating the index. */
|
||||
PaUtil_WriteMemoryBarrier();
|
||||
return rbuf->readIndex = (rbuf->readIndex + elementCount) & rbuf->bigMask;
|
||||
}
|
||||
|
||||
/***************************************************************************
|
||||
** Return elements written. */
|
||||
ring_buffer_size_t PaUtil_WriteRingBuffer( PaUtilRingBuffer *rbuf, const void *data, ring_buffer_size_t elementCount )
|
||||
{
|
||||
ring_buffer_size_t size1, size2, numWritten;
|
||||
void *data1, *data2;
|
||||
numWritten = PaUtil_GetRingBufferWriteRegions( rbuf, elementCount, &data1, &size1, &data2, &size2 );
|
||||
if( size2 > 0 )
|
||||
{
|
||||
|
||||
memcpy( data1, data, size1*rbuf->elementSizeBytes );
|
||||
data = ((char *)data) + size1*rbuf->elementSizeBytes;
|
||||
memcpy( data2, data, size2*rbuf->elementSizeBytes );
|
||||
}
|
||||
else
|
||||
{
|
||||
memcpy( data1, data, size1*rbuf->elementSizeBytes );
|
||||
}
|
||||
PaUtil_AdvanceRingBufferWriteIndex( rbuf, numWritten );
|
||||
return numWritten;
|
||||
}
|
||||
|
||||
/***************************************************************************
|
||||
** Return elements read. */
|
||||
ring_buffer_size_t PaUtil_ReadRingBuffer( PaUtilRingBuffer *rbuf, void *data, ring_buffer_size_t elementCount )
|
||||
{
|
||||
ring_buffer_size_t size1, size2, numRead;
|
||||
void *data1, *data2;
|
||||
numRead = PaUtil_GetRingBufferReadRegions( rbuf, elementCount, &data1, &size1, &data2, &size2 );
|
||||
if( size2 > 0 )
|
||||
{
|
||||
memcpy( data, data1, size1*rbuf->elementSizeBytes );
|
||||
data = ((char *)data) + size1*rbuf->elementSizeBytes;
|
||||
memcpy( data, data2, size2*rbuf->elementSizeBytes );
|
||||
}
|
||||
else
|
||||
{
|
||||
memcpy( data, data1, size1*rbuf->elementSizeBytes );
|
||||
}
|
||||
PaUtil_AdvanceRingBufferReadIndex( rbuf, numRead );
|
||||
return numRead;
|
||||
}
|
||||
@ -1,233 +0,0 @@
|
||||
#ifndef WEBRTC_AUDIO_DEVICE_PA_RINGBUFFER_H
|
||||
#define WEBRTC_AUDIO_DEVICE_PA_RINGBUFFER_H
|
||||
/*
|
||||
* $Id: pa_ringbuffer.h 1421 2009-11-18 16:09:05Z bjornroche $
|
||||
* Portable Audio I/O Library
|
||||
* Ring Buffer utility.
|
||||
*
|
||||
* Author: Phil Burk, http://www.softsynth.com
|
||||
* modified for SMP safety on OS X by Bjorn Roche.
|
||||
* also allowed for const where possible.
|
||||
* modified for multiple-byte-sized data elements by Sven Fischer
|
||||
*
|
||||
* Note that this is safe only for a single-thread reader
|
||||
* and a single-thread writer.
|
||||
*
|
||||
* This program is distributed with the PortAudio Portable Audio Library.
|
||||
* For more information see: http://www.portaudio.com
|
||||
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
|
||||
*
|
||||
* Permission is hereby granted, free of charge, to any person obtaining
|
||||
* a copy of this software and associated documentation files
|
||||
* (the "Software"), to deal in the Software without restriction,
|
||||
* including without limitation the rights to use, copy, modify, merge,
|
||||
* publish, distribute, sublicense, and/or sell copies of the Software,
|
||||
* and to permit persons to whom the Software is furnished to do so,
|
||||
* subject to the following conditions:
|
||||
*
|
||||
* The above copyright notice and this permission notice shall be
|
||||
* included in all copies or substantial portions of the Software.
|
||||
*
|
||||
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
|
||||
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
|
||||
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
|
||||
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
|
||||
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
|
||||
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
|
||||
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
|
||||
*/
|
||||
|
||||
/*
|
||||
* The text above constitutes the entire PortAudio license; however,
|
||||
* the PortAudio community also makes the following non-binding requests:
|
||||
*
|
||||
* Any person wishing to distribute modifications to the Software is
|
||||
* requested to send the modifications to the original developer so that
|
||||
* they can be incorporated into the canonical version. It is also
|
||||
* requested that these non-binding requests be included along with the
|
||||
* license above.
|
||||
*/
|
||||
|
||||
/** @file
|
||||
@ingroup common_src
|
||||
@brief Single-reader single-writer lock-free ring buffer
|
||||
|
||||
PaUtilRingBuffer is a ring buffer used to transport samples between
|
||||
different execution contexts (threads, OS callbacks, interrupt handlers)
|
||||
without requiring the use of any locks. This only works when there is
|
||||
a single reader and a single writer (ie. one thread or callback writes
|
||||
to the ring buffer, another thread or callback reads from it).
|
||||
|
||||
The PaUtilRingBuffer structure manages a ring buffer containing N
|
||||
elements, where N must be a power of two. An element may be any size
|
||||
(specified in bytes).
|
||||
|
||||
The memory area used to store the buffer elements must be allocated by
|
||||
the client prior to calling PaUtil_InitializeRingBuffer() and must outlive
|
||||
the use of the ring buffer.
|
||||
*/
|
||||
|
||||
#if defined(__APPLE__)
|
||||
#include <sys/types.h>
|
||||
typedef int32_t ring_buffer_size_t;
|
||||
#elif defined( __GNUC__ )
|
||||
typedef long ring_buffer_size_t;
|
||||
#elif (_MSC_VER >= 1400)
|
||||
typedef long ring_buffer_size_t;
|
||||
#elif defined(_MSC_VER) || defined(__BORLANDC__)
|
||||
typedef long ring_buffer_size_t;
|
||||
#else
|
||||
typedef long ring_buffer_size_t;
|
||||
#endif
|
||||
|
||||
|
||||
|
||||
#ifdef __cplusplus
|
||||
extern "C"
|
||||
{
|
||||
#endif /* __cplusplus */
|
||||
|
||||
typedef struct PaUtilRingBuffer
|
||||
{
|
||||
ring_buffer_size_t bufferSize; /**< Number of elements in FIFO. Power of 2. Set by PaUtil_InitRingBuffer. */
|
||||
ring_buffer_size_t writeIndex; /**< Index of next writable element. Set by PaUtil_AdvanceRingBufferWriteIndex. */
|
||||
ring_buffer_size_t readIndex; /**< Index of next readable element. Set by PaUtil_AdvanceRingBufferReadIndex. */
|
||||
ring_buffer_size_t bigMask; /**< Used for wrapping indices with extra bit to distinguish full/empty. */
|
||||
ring_buffer_size_t smallMask; /**< Used for fitting indices to buffer. */
|
||||
ring_buffer_size_t elementSizeBytes; /**< Number of bytes per element. */
|
||||
char *buffer; /**< Pointer to the buffer containing the actual data. */
|
||||
}PaUtilRingBuffer;
|
||||
|
||||
/** Initialize Ring Buffer.
|
||||
|
||||
@param rbuf The ring buffer.
|
||||
|
||||
@param elementSizeBytes The size of a single data element in bytes.
|
||||
|
||||
@param elementCount The number of elements in the buffer (must be power of 2).
|
||||
|
||||
@param dataPtr A pointer to a previously allocated area where the data
|
||||
will be maintained. It must be elementCount*elementSizeBytes long.
|
||||
|
||||
@return -1 if elementCount is not a power of 2, otherwise 0.
|
||||
*/
|
||||
ring_buffer_size_t PaUtil_InitializeRingBuffer( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementSizeBytes, ring_buffer_size_t elementCount, void *dataPtr );
|
||||
|
||||
/** Clear buffer. Should only be called when buffer is NOT being read.
|
||||
|
||||
@param rbuf The ring buffer.
|
||||
*/
|
||||
void PaUtil_FlushRingBuffer( PaUtilRingBuffer *rbuf );
|
||||
|
||||
/** Retrieve the number of elements available in the ring buffer for writing.
|
||||
|
||||
@param rbuf The ring buffer.
|
||||
|
||||
@return The number of elements available for writing.
|
||||
*/
|
||||
ring_buffer_size_t PaUtil_GetRingBufferWriteAvailable( PaUtilRingBuffer *rbuf );
|
||||
|
||||
/** Retrieve the number of elements available in the ring buffer for reading.
|
||||
|
||||
@param rbuf The ring buffer.
|
||||
|
||||
@return The number of elements available for reading.
|
||||
*/
|
||||
ring_buffer_size_t PaUtil_GetRingBufferReadAvailable( PaUtilRingBuffer *rbuf );
|
||||
|
||||
/** Write data to the ring buffer.
|
||||
|
||||
@param rbuf The ring buffer.
|
||||
|
||||
@param data The address of new data to write to the buffer.
|
||||
|
||||
@param elementCount The number of elements to be written.
|
||||
|
||||
@return The number of elements written.
|
||||
*/
|
||||
ring_buffer_size_t PaUtil_WriteRingBuffer( PaUtilRingBuffer *rbuf, const void *data, ring_buffer_size_t elementCount );
|
||||
|
||||
/** Read data from the ring buffer.
|
||||
|
||||
@param rbuf The ring buffer.
|
||||
|
||||
@param data The address where the data should be stored.
|
||||
|
||||
@param elementCount The number of elements to be read.
|
||||
|
||||
@return The number of elements read.
|
||||
*/
|
||||
ring_buffer_size_t PaUtil_ReadRingBuffer( PaUtilRingBuffer *rbuf, void *data, ring_buffer_size_t elementCount );
|
||||
|
||||
/** Get address of region(s) to which we can write data.
|
||||
|
||||
@param rbuf The ring buffer.
|
||||
|
||||
@param elementCount The number of elements desired.
|
||||
|
||||
@param dataPtr1 The address where the first (or only) region pointer will be
|
||||
stored.
|
||||
|
||||
@param sizePtr1 The address where the first (or only) region length will be
|
||||
stored.
|
||||
|
||||
@param dataPtr2 The address where the second region pointer will be stored if
|
||||
the first region is too small to satisfy elementCount.
|
||||
|
||||
@param sizePtr2 The address where the second region length will be stored if
|
||||
the first region is too small to satisfy elementCount.
|
||||
|
||||
@return The room available to be written or elementCount, whichever is smaller.
|
||||
*/
|
||||
ring_buffer_size_t PaUtil_GetRingBufferWriteRegions( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount,
|
||||
void **dataPtr1, ring_buffer_size_t *sizePtr1,
|
||||
void **dataPtr2, ring_buffer_size_t *sizePtr2 );
|
||||
|
||||
/** Advance the write index to the next location to be written.
|
||||
|
||||
@param rbuf The ring buffer.
|
||||
|
||||
@param elementCount The number of elements to advance.
|
||||
|
||||
@return The new position.
|
||||
*/
|
||||
ring_buffer_size_t PaUtil_AdvanceRingBufferWriteIndex( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount );
|
||||
|
||||
/** Get address of region(s) from which we can write data.
|
||||
|
||||
@param rbuf The ring buffer.
|
||||
|
||||
@param elementCount The number of elements desired.
|
||||
|
||||
@param dataPtr1 The address where the first (or only) region pointer will be
|
||||
stored.
|
||||
|
||||
@param sizePtr1 The address where the first (or only) region length will be
|
||||
stored.
|
||||
|
||||
@param dataPtr2 The address where the second region pointer will be stored if
|
||||
the first region is too small to satisfy elementCount.
|
||||
|
||||
@param sizePtr2 The address where the second region length will be stored if
|
||||
the first region is too small to satisfy elementCount.
|
||||
|
||||
@return The number of elements available for reading.
|
||||
*/
|
||||
ring_buffer_size_t PaUtil_GetRingBufferReadRegions( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount,
|
||||
void **dataPtr1, ring_buffer_size_t *sizePtr1,
|
||||
void **dataPtr2, ring_buffer_size_t *sizePtr2 );
|
||||
|
||||
/** Advance the read index to the next location to be read.
|
||||
|
||||
@param rbuf The ring buffer.
|
||||
|
||||
@param elementCount The number of elements to advance.
|
||||
|
||||
@return The new position.
|
||||
*/
|
||||
ring_buffer_size_t PaUtil_AdvanceRingBufferReadIndex( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount );
|
||||
|
||||
#ifdef __cplusplus
|
||||
}
|
||||
#endif /* __cplusplus */
|
||||
#endif /* MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_PORTAUDIO_PA_RINGBUFFER_H_ */
|
||||
Loading…
Reference in New Issue
Block a user