AudioIO.h

/**********************************************************************

  Audacity: A Digital Audio Editor

  AudioIO.h

  Dominic Mazzoni

  Use the PortAudio library to play and record sound

**********************************************************************/

#ifndef __AUDACITY_AUDIO_IO__
#define __AUDACITY_AUDIO_IO__

#include "portaudio.h"
#include "Audacity.h"
#include "Experimental.h"

#ifdef USE_MIDI
#ifdef EXPERIMENTAL_MIDI_OUT
#include "portmidi.h"
#include "porttime.h"
#include "allegro.h"
#endif // EXPERIMENTAL_MIDI_OUT
#endif // USE_MIDI

#if USE_PORTMIXER
#include "portmixer.h"
#endif

#include <wx/string.h>
#include <wx/thread.h>

#include "WaveTrack.h"
#include "SampleFormat.h"

class AudioIO;
class RingBuffer;
class Mixer;
class Resample;
class TimeTrack;
class AudioThread;
class Meter;
class TimeTrack;
class wxDialog;

extern AUDACITY_DLL_API AudioIO *gAudioIO;

void InitAudioIO();
void DeinitAudioIO();
wxString DeviceName(const PaDeviceInfo* info);
wxString HostName(const PaDeviceInfo* info);
bool ValidateDeviceNames();

class AUDACITY_DLL_API AudioIOListener {
public:
   AudioIOListener() {}
   virtual ~AudioIOListener() {}
   
   virtual void OnAudioIORate(int rate) = 0;
   virtual void OnAudioIOStartRecording() = 0;
   virtual void OnAudioIOStopRecording() = 0;
   virtual void OnAudioIONewBlockFiles(const wxString& blockFileLog) = 0;
};

#define BAD_STREAM_TIME -1000000000.0

#define MAX_MIDI_BUFFER_SIZE 5000
#define DEFAULT_SYNTH_LATENCY 5

#define DEFAULT_LATENCY_DURATION 100.0
#define DEFAULT_LATENCY_CORRECTION -130.0

#ifdef AUTOMATED_INPUT_LEVEL_ADJUSTMENT
   #define AILA_DEF_TARGET_PEAK 92
   #define AILA_DEF_DELTA_PEAK 2
   #define AILA_DEF_ANALYSIS_TIME 1000
   #define AILA_DEF_NUMBER_ANALYSIS 5
#endif

class AUDACITY_DLL_API AudioIO {

 public:
   AudioIO();
   ~AudioIO();

   AudioIOListener* GetListener() { return mListener; }
   void SetListener(AudioIOListener* listener);
   
   /** \brief Start up Portaudio for capture and recording as needed for
    * input monitoring and software playthrough only
    *
    * This uses the Default project sample format, current sample rate, and 
    * selected number of input channels to open the recording device and start
    * reading input data. If software playthrough is enabled, it also opens
    * the output device in stereo to play the data through */
   void StartMonitoring(double sampleRate);

   /** \brief Start recording or playing back audio
    *
    * Allocates buffers for recording and playback, gets the Audio thread to
    * fill them, and sets the stream rolling.
    * If successful, returns a token identifying this particular stream
    * instance.  For use with IsStreamActive() below */

   int StartStream(WaveTrackArray playbackTracks, WaveTrackArray captureTracks,
#ifdef EXPERIMENTAL_MIDI_OUT
                   NoteTrackArray midiTracks,
#endif
                   TimeTrack *timeTrack, double sampleRate,
                   double t0, double t1,
                   AudioIOListener* listener,
                   bool playLooped = false,
                   double cutPreviewGapStart = 0.0,
                   double cutPreviewGapLen = 0.0);

   /** \brief Stop recording, playback or input monitoring.
    *
    * Does quite a bit of housekeeping, including switching off monitoring,
    * flushing recording buffers out to wave tracks, and applies latency
    * correction to recorded tracks if necessary */
   void StopStream();
   /** \brief Move the playback / recording position of the current stream
    * by the specified amount from where it is now */
   void SeekStream(double seconds) { mSeek = seconds; };
   
   /** \brief  Returns true if audio i/o is busy starting, stopping, playing,
    * or recording.
    *
    * When this is false, it's safe to start playing or recording */
   bool IsBusy();

   /** \brief Returns true if the audio i/o is running at all, but not during
    * cleanup
    *
    * Doesn't return true if the device has been closed but some disk i/o or
    * cleanup is still going on. If you want to know if it's safe to start a
    * new stream, use IsBusy() */
   bool IsStreamActive();
   bool IsStreamActive(int token);

#ifdef EXPERIMENTAL_MIDI_OUT
   /** \brief Compute the current PortMidi timestamp time.
    * 
    * This is used by PortMidi to synchronize midi time to audio samples
    */
   PmTimestamp MidiTime();

   // Note: audio code solves the problem of soloing/muting tracks by scanning
   // all playback tracks on every call to the audio buffer fill routine.
   // We do the same for Midi, but it seems wasteful for at least two
   // threads to be frequently polling to update status. This could be
   // eliminated (also with a reduction in code I think) by updating mHasSolo
   // each time a solo button is activated or deactivated. For now, I'm 
   // going to do this polling in the FillMidiBuffer routine to localize
   // changes for midi to the midi code, but I'm declaring the variable
   // here so possibly in the future, Audio code can use it too. -RBD
 private:
   bool  mHasSolo; // is any playback solo button pressed?
 public:
   bool SetHasSolo(bool hasSolo); 
   bool GetHasSolo() { return mHasSolo; }
   void SetMidiPlaySpeed(double s) { mMidiPlaySpeed = s * 0.01; }
#endif

   /** \brief Returns true if the stream is active, or even if audio I/O is
    * busy cleaning up its data or writing to disk.
    *
    * This is used by TrackPanel to determine when a track has been completely
    * recorded, and it's safe to flush to disk. */
   bool IsAudioTokenActive(int token);

   /** \brief Returns true if we're monitoring input (but not recording or 
    * playing actual audio) */
   bool IsMonitoring();

   /** \brief Pause and un-pause playback and recording */
   void SetPaused(bool state);
   /** \brief Find out if playback / recording is currently paused */
   bool IsPaused();

   /* Mixer services are always available.  If no stream is running, these
    * methods use whatever device is specified by the preferences.  If a
    * stream *is* running, naturally they manipulate the mixer associated
    * with that stream.  If no mixer is available, output is emulated and 
    * input is stuck at 1.0f (a gain is applied to output samples).
    */
   void SetMixer(int inputSource);
   void SetMixer(int inputSource, float inputVolume,
                 float playbackVolume);
   void GetMixer(int *inputSource, float *inputVolume,
                 float *playbackVolume);
   /** @brief Find out if the input hardware level control is available
    *
    * Checks the mInputMixerWorks variable, which is set up in
    * AudioIO::HandleDeviceChange(). External people care, because we want to
    * disable the UI if it doesn't work.
    */
   bool InputMixerWorks();

   /** @brief Find out if the output level control is being emulated via software attenuation
    *
    * Checks the mEmulateMixerOutputVol variable, which is set up in
    * AudioIO::HandleDeviceChange(). External classes care, because we want to
    * modify the UI if it doesn't work.
    */
   bool OutputMixerEmulated();

   /** \brief Get the list of inputs to the current mixer device
    *
    * Returns an array of strings giving the names of the inputs to the 
    * soundcard mixer (driven by PortMixer) */
   wxArrayString GetInputSourceNames();

   /** \brief update state after changing what audio devices are selected 
    *
    * Called when the devices stored in the preferences are changed to update
    * the audio mixer capabilities
    *
    * \todo: Make this do a sample rate query and store the result in the
    * AudioIO object to avoid doing it later? Would simplify the
    * GetSupported*Rate functions considerably */
   void HandleDeviceChange();

   /** \brief Set the current VU meters - this should be done once after 
    * each call to StartStream currently */
   void SetMeters(Meter *inputMeter, Meter *outputMeter);
   
   /** \brief Get a list of sample rates the output (playback) device
    * supports.
    *
    * If no information about available sample rates can be fetched,
    * an empty list is returned.
    *
    * You can explicitely give the index of the device.  If you don't
    * give it, the currently selected device from the preferences will be used.
    *
    * You may also specify a rate for which to check in addition to the
    * standard rates.
    */
   static wxArrayLong GetSupportedPlaybackRates(int DevIndex = -1,
                                                double rate = 0.0);

   /** \brief Get a list of sample rates the input (recording) device
    * supports.
    *
    * If no information about available sample rates can be fetched,
    * an empty list is returned.
    *
    * You can explicitely give the index of the device.  If you don't
    * give it, the currently selected device from the preferences will be used.
    *
    * You may also specify a rate for which to check in addition to the
    * standard rates.
    */
   static wxArrayLong GetSupportedCaptureRates(int devIndex = -1,
                                               double rate = 0.0);

   /** \brief Get a list of sample rates the current input/output device
    * combination supports.
    *
    * Since there is no concept (yet) for different input/output
    * sample rates, this currently returns only sample rates that are
    * supported on both the output and input device. If no information
    * about available sample rates can be fetched, it returns a default
    * list.
    * You can explicitely give the indexes of the playDevice/recDevice.
    * If you don't give them, the selected devices from the preferences
    * will be used.
    * You may also specify a rate for which to check in addition to the
    * standard rates.
    */
   static wxArrayLong GetSupportedSampleRates(int playDevice = -1,
                                              int recDevice = -1,
                                       double rate = 0.0);

   /** \brief Get a supported sample rate which can be used a an optimal
    * default.
    *
    * Currently, this uses the first supported rate in the list
    * [44100, 48000, highest sample rate]. Used in Project as a default value
    * for project rates if one cannot be retrieved from the preferences.
    * So all in all not that useful or important really
    */
   static int GetOptimalSupportedSampleRate();

   /** \brief The time the stream has been playing for
    *
    * This is given in seconds based on starting at t0
    * When playing looped, this will start from t0 again,
    * too. So the returned time should be always between
    * t0 and t1
    */
   double GetStreamTime();

   sampleFormat GetCaptureFormat() { return mCaptureFormat; }
   int GetNumCaptureChannels() { return mNumCaptureChannels; }

   /** \brief Array of common audio sample rates
    *
    * These are the rates we will always support, regardless of hardware support
    * for them (by resampling in audacity if needed) */
   static const int StandardRates[];
   /** \brief How many standard sample rates there are */
   static const int NumStandardRates;

   /** \brief Get diagnostic information on all the available audio I/O devices
    *
    */
   wxString GetDeviceInfo();

   /** \brief Ensure selected device names are valid
    *
    */
   static bool ValidateDeviceNames(wxString play, wxString rec);

   /** \brief Function to automatically set an acceptable volume
    *
    */
   #ifdef AUTOMATED_INPUT_LEVEL_ADJUSTMENT
      void AILAInitialize();
      void AILADisable();
      bool AILAIsActive();
      void AILAProcess(double maxPeak);
      void AILASetStartTime();
      double AILAGetLastDecisionTime();
   #endif

private:
   /** \brief Return a valid sample rate that is supported by the current I/O
    * device(s).
    *
    * The return from this function is used to determine the sample rate that
    * audacity actually runs the audio I/O stream at. if there is no suitable
    * rate available from the hardware, it returns 0.
    * The sampleRate argument gives the desired sample rate (the rate of the
    * audio to be handeled, i.e. the currently Project Rate).
    * capturing is true if the stream is capturing one or more audio channels,
    * and playing is true if one or more channels are being played. */
   double GetBestRate(bool capturing, bool playing, double sampleRate);

   /** \brief Opens the portaudio stream(s) used to do playback or recording
    * (or both) through.
    *
    * The sampleRate passed is the Project Rate of the active project. It may
    * or may not be actually supported by playback or recording hardware
    * currently in use (for many reasons). The number of Capture and Playback
    * channels requested includes an allocation for doing software playthrough
    * if necessary. The captureFormat is used for recording only, the playback
    * being floating point always. Returns true if the stream opened sucessfully
    * and false if it did not. */
   bool StartPortAudioStream(double sampleRate,
                             unsigned int numPlaybackChannels,
                             unsigned int numCaptureChannels,
                             sampleFormat captureFormat);
   void FillBuffers();

#ifdef EXPERIMENTAL_MIDI_OUT
   void PrepareMidiIterator(bool send = true, double offset = 0);
   bool StartPortMidiStream();
   void OutputEvent();
   void FillMidiBuffers();
   void GetNextEvent();
   void AudacityMidiCallback();
   double AudioTime() { return mT0 + mNumFrames / mRate; }
   double PauseTime();
   // double getCurrentTrackTime();
   // long CalculateMidiTimeStamp(double time);
   void AllNotesOff();
#endif

   /** \brief Get the number of audio samples free in all of the playback
    * buffers.
    *
    * Returns the smallest of the buffer free space values in the event that
    * they are different. */
   int GetCommonlyAvailPlayback();

   /** \brief Get the number of audio samples ready in all of the recording
    * buffers.
    *
    * Returns the smallest of the number of samples available for storage in 
    * the recording buffers (i.e. the number of samples that can be read from
    * all record buffers without underflow). */
   int GetCommonlyAvailCapture();

   /** \brief get the index of the supplied (named) recording device, or the
    * device selected in the preferences if none given.
    *
    * Pure utility function, but it comes round a number of times in the code
    * and would be neater done once. If the device isn't found, return the
    * default device index.
    */
   static int getRecordDevIndex(wxString devName = wxT(""));
   /** \brief get the index of the device selected in the preferences.
    *
    * If the device isn't found, returns -1
    */
#if USE_PORTMIXER
   static int getRecordSourceIndex(PxMixer *portMixer);
#endif

   /** \brief get the index of the supplied (named) playback device, or the
    * device selected in the preferences if none given.
    *
    * Pure utility function, but it comes round a number of times in the code
    * and would be neater done once. If the device isn't found, return the
    * default device index.
    */
   static int getPlayDevIndex(wxString devName = wxT(""));

   /** \brief Array of audio sample rates to try to use
    *
    * These are the rates we will check if a device supports, and is as long
    * as I can think of (to try and work out what the card can do) */
   static const int RatesToTry[];
   /** \brief How many sample rates to try */
   static const int NumRatesToTry;

   double NormalizeStreamTime(double absoluteTime) const;

   /** \brief Clean up after StartStream if it fails.
     *
     * If bOnlyBuffers is specified, it only cleans up the buffers. */
   void StartStreamCleanup(bool bOnlyBuffers = false);

#ifdef EXPERIMENTAL_MIDI_OUT
   //   MIDI_PLAYBACK:
   PmStream        *mMidiStream;
   PmError          mLastPmError;
   long             mMidiLatency; // latency value for PortMidi
   long             mSynthLatency; // latency of MIDI synthesizer
   double           mMidiPlaySpeed; // a copy of TranscriptionToolBar::mPlaySpeed

   // These fields are used to synchronize MIDI with audio
   volatile double  mAudioCallbackOutputTime; // PortAudio's outTime
   volatile long    mNumFrames;         // includes pauses
   volatile long    mNumPauseFrames;    // how many frames of zeros inserted?
   volatile long    mPauseTime;         // pause in ms if no audio playback
   volatile double  mMidiLoopOffset;    // total of backward jumps
   volatile long    mAudioFramesPerBuffer;
   volatile bool    mMidiPaused;        // used by Midi process to record
       // that pause has begun. Pause time is accumulated in mPauseTime.
       // This variable is shared so that it can be cleared when playback
       // begins.

   Alg_seq_ptr      mSeq;
   Alg_iterator_ptr mIterator;
   Alg_event_ptr    mNextEvent; // the next event to play (or null)
   double           mNextEventTime; // the time of the next event
                       // (note that this could be a note's time+duration)
   NoteTrack        *mNextEventTrack; // track of next event
   bool             mMidiOutputComplete; // true when output reaches mT1
   bool             mNextIsNoteOn; // is the next event a note-off?
   //   int                 mCnt;
   // mMidiStreamActive tells when mMidiStream is open for output
   bool             mMidiStreamActive;
   // when true, mSendMidiState means send only updates, not note-on's,
   // used to send state changes that precede the selected notes
   bool             mSendMidiState;
   NoteTrackArray   mMidiPlaybackTracks;
#endif

#ifdef AUTOMATED_INPUT_LEVEL_ADJUSTMENT
   bool           mAILAActive;
   bool           mAILAClipped;
   int            mAILATotalAnalysis;
   int            mAILAAnalysisCounter;
   double         mAILAMax;
   double         mAILAGoalPoint;
   double         mAILAGoalDelta;
   double         mAILAAnalysisTime;
   double         mAILALastStartTime;
   double         mAILAChangeFactor;
   double         mAILATopLevel;
   double         mAILAAnalysisEndTime;
   double         mAILAAbsolutStartTime;
   unsigned short mAILALastChangeType;  //0 - no change, 1 - increase change, 2 - decrease change
#endif

   AudioThread        *mThread;
#ifdef EXPERIMENTAL_MIDI_OUT
   AudioThread         *mMidiThread;
#endif
   Resample          **mResample;
   RingBuffer        **mCaptureBuffers;
   WaveTrackArray      mCaptureTracks;
   RingBuffer        **mPlaybackBuffers;
   WaveTrackArray      mPlaybackTracks;

   Mixer             **mPlaybackMixers;
   int                 mStreamToken;
   int                 mStopStreamCount;
   static int          mNextStreamToken;
   double              mFactor;
   double              mRate;
   double              mT0; // playback starts at offset of mT0
   double              mT1; // and ends at offset of mT1
   double              mTime; // current time position during playback
   double              mWarpedTime; // current time after warping, starting at zero (unlike mTime)
   double              mWarpedLength; // total length after warping
   double              mSeek;
   double              mPlaybackRingBufferSecs;
   double              mCaptureRingBufferSecs;
   double              mMaxPlaybackSecsToCopy;
   double              mMinCaptureSecsToCopy;
   bool                mPaused;
   PaStream           *mPortStreamV19;
   bool                mSoftwarePlaythrough;
   bool                mPauseRec;
   float               mSilenceLevel;
   unsigned int        mNumCaptureChannels;
   unsigned int        mNumPlaybackChannels;
   sampleFormat        mCaptureFormat;
   int                 mLostSamples;
   volatile bool       mAudioThreadShouldCallFillBuffersOnce;
   volatile bool       mAudioThreadFillBuffersLoopRunning;
   volatile bool       mAudioThreadFillBuffersLoopActive;

#ifdef EXPERIMENTAL_MIDI_OUT
   volatile bool       mMidiThreadFillBuffersLoopRunning;
   volatile bool       mMidiThreadFillBuffersLoopActive;
#endif

   volatile double     mLastRecordingOffset;
   PaError             mLastPaError;

   Meter              *mInputMeter;
   Meter              *mOutputMeter;
   bool                mUpdateMeters;
   bool                mUpdatingMeters;

   #if USE_PORTMIXER
   PxMixer            *mPortMixer;
   float               mPreviousHWPlaythrough;
   #endif /* USE_PORTMIXER */

   bool                mEmulateMixerOutputVol;
   /** @brief Can we control the hardware input level?
    *
    * This flag is set to true if using portmixer to control the 
    * input volume seems to be working (and so we offer the user the control),
    * and to false (locking the control out) otherwise. This avoids stupid
    * scaled clipping problems when trying to do software emulated input volume
    * control */
   bool                mInputMixerWorks;
   float               mMixerOutputVol;

   bool                mPlayLooped;
   double              mCutPreviewGapStart;
   double              mCutPreviewGapLen;
   
   samplePtr mSilentBuf;
   sampleCount mLastSilentBufSize;
   
   AudioIOListener*    mListener;

   friend class AudioThread;
#ifdef EXPERIMENTAL_MIDI_OUT
   friend class MidiThread;
#endif

   friend void InitAudioIO();
   friend void DeinitAudioIO();

   TimeTrack *mTimeTrack;

   // For cacheing supported sample rates
   static int mCachedPlaybackIndex;
   static wxArrayLong mCachedPlaybackRates;
   static int mCachedCaptureIndex;
   static wxArrayLong mCachedCaptureRates;
   static wxArrayLong mCachedSampleRates;
   static double mCachedBestRateIn;
   static double mCachedBestRateOut;

   /** brief The function which is called from PortAudio's callback thread
    * context to collect and deliver audio for / from the sound device.
    *
    * This covers recording, playback, and doing both simultaneously. It is
    * also invoked to do monitoring and software playthrough. Note that dealing
    * with the two buffers needs some care to ensure that the right things
    * happen for all possible cases.
    * @param inputBuffer Buffer of length framesPerBuffer containing samples
    * from the sound card, or null if not capturing audio. Note that the data
    * type will depend on the format of audio data that was chosen when the
    * stream was created (so could be floats or various integers)
    * @param outputBuffer Uninitialised buffer of length framesPerBuffer which
    * will be sent to the sound card after the callback, or null if not playing
    * audio back.
    * @param framesPerBuffer The length of the playback and recording buffers
    * @param PaStreamCallbackTimeInfo Pointer to PortAudio time information
    * structure, which tells us how long we have been playing / recording
    * @param statusFlags PortAudio stream status flags
    * @param userData pointer to user-defined data structure. Provided for
    * flexibility by PortAudio, but not used by Audacity - the data is stored in
    * the AudioIO class instead.
    */
   friend int audacityAudioCallback(
                const void *inputBuffer, void *outputBuffer,
                unsigned long framesPerBuffer,
                const PaStreamCallbackTimeInfo *timeInfo,
                PaStreamCallbackFlags statusFlags, void *userData );
};

#endif