Duplicate AudioIOBase into PlaybackSchedule...

... please verify empty file diffs; this simplifies review of the next commit

src/PlaybackSchedule.h

@@ -0,0 +1,486 @@
+/**********************************************************************
+
+Audacity: A Digital Audio Editor
+
+AudioIOBase.h
+
+Paul Licameli split from AudioIO.h
+
+**********************************************************************/
+
+#ifndef __AUDACITY_AUDIO_IO_BASE__
+#define __AUDACITY_AUDIO_IO_BASE__
+
+
+
+
+#include <atomic>
+#include <cfloat>
+#include <functional>
+#include <memory>
+#include <vector>
+#include <wx/string.h>
+#include <wx/weakref.h> // member variable
+
+struct PaDeviceInfo;
+typedef void PaStream;
+
+#if USE_PORTMIXER
+typedef void PxMixer;
+#endif
+
+class AudioIOBase;
+
+class AudacityProject;
+class AudioIOListener;
+class BoundedEnvelope;
+// Windows build needs complete type for parameter of wxWeakRef
+// class MeterPanelBase;
+#include "widgets/MeterPanelBase.h"
+using PRCrossfadeData = std::vector< std::vector < float > >;
+
+#define BAD_STREAM_TIME (-DBL_MAX)
+
+// For putting an increment of work in the scrubbing queue
+struct ScrubbingOptions {
+   ScrubbingOptions() {}
+
+   bool adjustStart {};
+
+   // usually from TrackList::GetEndTime()
+   double maxTime {};
+   double minTime {};
+
+   bool bySpeed {};
+   bool isPlayingAtSpeed{};
+   bool isKeyboardScrubbing{};
+
+   double delay {};
+
+   // Initial and limiting values for the speed of a scrub interval:
+   double initSpeed { 1.0 };
+   double minSpeed { 0.0 };
+   double maxSpeed { 1.0 };
+
+
+   // When maximum speed scrubbing skips to follow the mouse,
+   // this is the minimum amount of playback allowed at the maximum speed:
+   double minStutterTime {};
+
+   static double MaxAllowedScrubSpeed()
+   { return 32.0; } // Is five octaves enough for your amusement?
+   static double MinAllowedScrubSpeed()
+   { return 0.01; } // Mixer needs a lower bound speed.  Scrub no slower than this.
+};
+
+// To avoid growing the argument list of StartStream, add fields here
+struct AudioIOStartStreamOptions
+{
+   explicit
+   AudioIOStartStreamOptions(AudacityProject *pProject_, double rate_)
+      : pProject{ pProject_ }
+      , envelope(nullptr)
+      , rate(rate_)
+      , playLooped(false)
+      , cutPreviewGapStart(0.0)
+      , cutPreviewGapLen(0.0)
+      , pStartTime(NULL)
+      , preRoll(0.0)
+   {}
+
+   AudacityProject *pProject{};
+   MeterPanelBase *captureMeter{}, *playbackMeter{};
+   const BoundedEnvelope *envelope; // for time warping
+   std::shared_ptr< AudioIOListener > listener;
+   double rate;
+   bool playLooped;
+   double cutPreviewGapStart;
+   double cutPreviewGapLen;
+   double * pStartTime;
+   double preRoll;
+
+#ifdef EXPERIMENTAL_SCRUBBING_SUPPORT
+   // Non-null value indicates that scrubbing will happen
+   // (do not specify a time track, looping, or recording, which
+   //  are all incompatible with scrubbing):
+   ScrubbingOptions *pScrubbingOptions {};
+#endif
+
+   // contents may get swapped with empty vector
+   PRCrossfadeData      *pCrossfadeData{};
+
+   // An unfortunate thing needed just to make scrubbing work on Linux when
+   // we can't use a separate polling thread.
+   // The return value is a number of milliseconds to sleep before calling again
+   std::function< unsigned long() > playbackStreamPrimer;
+};
+
+///\brief A singleton object supporting queries of the state of any active
+/// audio streams, and audio device capabilities
+class AUDACITY_DLL_API AudioIOBase /* not final */
+{
+public:
+   static AudioIOBase *Get();
+
+   virtual ~AudioIOBase();
+
+   void SetCaptureMeter(AudacityProject *project, MeterPanelBase *meter);
+   void SetPlaybackMeter(AudacityProject *project, MeterPanelBase *meter);
+
+   /** \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 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 explicitly 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 std::vector<long> 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 explicitly 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 std::vector<long> 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 explicitly 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 std::vector<long> 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 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();
+
+#ifdef EXPERIMENTAL_MIDI_OUT
+   /** \brief Get diagnostic information on all the available MIDI I/O devices */
+   wxString GetMidiDeviceInfo();
+#endif
+
+   /** \brief Find out if playback / recording is currently paused */
+   bool IsPaused() const;
+
+   virtual void StopStream() = 0;
+
+   /** \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() const;
+
+   /** \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() const;
+   bool IsStreamActive(int token) const;
+
+   /** \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) const;
+
+   /** \brief Returns true if we're monitoring input (but not recording or
+    * playing actual audio) */
+   bool IsMonitoring() const;
+
+   /* 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);
+
+protected:
+   static std::unique_ptr<AudioIOBase> ugAudioIO;
+   static wxString DeviceName(const PaDeviceInfo* info);
+   static wxString HostName(const PaDeviceInfo* info);
+
+   AudacityProject    *mOwningProject;
+
+   /// True if audio playback is paused
+   bool                mPaused;
+
+   /// True when output reaches mT1
+   bool             mMidiOutputComplete{ true };
+
+   /// mMidiStreamActive tells when mMidiStream is open for output
+   bool             mMidiStreamActive;
+
+   volatile int        mStreamToken;
+
+   /// Audio playback rate in samples per second
+   double              mRate;
+
+   PaStream           *mPortStreamV19;
+
+   wxWeakRef<MeterPanelBase> mInputMeter{};
+   wxWeakRef<MeterPanelBase> mOutputMeter{};
+
+   #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;
+
+   // For cacheing supported sample rates
+   static int mCachedPlaybackIndex;
+   static std::vector<long> mCachedPlaybackRates;
+   static int mCachedCaptureIndex;
+   static std::vector<long> mCachedCaptureRates;
+   static std::vector<long> mCachedSampleRates;
+   static double mCachedBestRateIn;
+
+   struct RecordingSchedule {
+      double mPreRoll{};
+      double mLatencyCorrection{}; // negative value usually
+      double mDuration{};
+      PRCrossfadeData mCrossfadeData;
+
+      // These are initialized by the main thread, then updated
+      // only by the thread calling FillBuffers:
+      double mPosition{};
+      bool mLatencyCorrected{};
+
+      double TotalCorrection() const { return mLatencyCorrection - mPreRoll; }
+      double ToConsume() const;
+      double Consumed() const;
+      double ToDiscard() const;
+   };
+
+   struct PlaybackSchedule {
+      /// Playback starts at offset of mT0, which is measured in seconds.
+      double              mT0;
+      /// Playback ends at offset of mT1, which is measured in seconds.  Note that mT1 may be less than mT0 during scrubbing.
+      double              mT1;
+      /// Current track time position during playback, in seconds.
+      /// Initialized by the main thread but updated by worker threads during
+      /// playback or recording, and periodically reread by the main thread for
+      /// purposes such as display update.
+      std::atomic<double> mTime;
+
+      /// Accumulated real time (not track position), starting at zero (unlike
+      /// mTime), and wrapping back to zero each time around looping play.
+      /// Thus, it is the length in real seconds between mT0 and mTime.
+      double              mWarpedTime;
+
+      /// Real length to be played (if looping, for each pass) after warping via a
+      /// time track, computed just once when starting the stream.
+      /// Length in real seconds between mT0 and mT1.  Always positive.
+      double              mWarpedLength;
+
+      // mWarpedTime and mWarpedLength are irrelevant when scrubbing,
+      // else they are used in updating mTime,
+      // and when not scrubbing or playing looped, mTime is also used
+      // in the test for termination of playback.
+
+      // with ComputeWarpedLength, it is now possible the calculate the warped length with 100% accuracy
+      // (ignoring accumulated rounding errors during playback) which fixes the 'missing sound at the end' bug
+      
+      const BoundedEnvelope *mEnvelope;
+      
+      volatile enum {
+         PLAY_STRAIGHT,
+         PLAY_LOOPED,
+#ifdef EXPERIMENTAL_SCRUBBING_SUPPORT
+         PLAY_SCRUB,
+         PLAY_AT_SPEED, // a version of PLAY_SCRUB.
+         PLAY_KEYBOARD_SCRUB,
+#endif
+      }                   mPlayMode { PLAY_STRAIGHT };
+      double              mCutPreviewGapStart;
+      double              mCutPreviewGapLen;
+
+      void Init(
+         double t0, double t1,
+         const AudioIOStartStreamOptions &options,
+         const RecordingSchedule *pRecordingSchedule );
+
+      /** \brief True if the end time is before the start time */
+      bool ReversedTime() const
+      {
+         return mT1 < mT0;
+      }
+
+      /** \brief Get current track time value, unadjusted
+       *
+       * Returns a time in seconds.
+       */
+      double GetTrackTime() const
+      { return mTime.load(std::memory_order_relaxed); }
+
+      /** \brief Set current track time value, unadjusted
+       */
+      void SetTrackTime( double time )
+      { mTime.store(time, std::memory_order_relaxed); }
+
+      /** \brief Clamps argument to be between mT0 and mT1
+       *
+       * Returns the bound if the value is out of bounds; does not wrap.
+       * Returns a time in seconds.
+       */
+      double ClampTrackTime( double trackTime ) const;
+
+      /** \brief Clamps mTime to be between mT0 and mT1
+       *
+       * Returns the bound if the value is out of bounds; does not wrap.
+       * Returns a time in seconds.
+       */
+      double LimitTrackTime() const;
+
+      /** \brief Normalizes mTime, clamping it and handling gaps from cut preview.
+       *
+       * Clamps the time (unless scrubbing), and skips over the cut section.
+       * Returns a time in seconds.
+       */
+      double NormalizeTrackTime() const;
+
+      void ResetMode() { mPlayMode = PLAY_STRAIGHT; }
+
+      bool PlayingStraight() const { return mPlayMode == PLAY_STRAIGHT; }
+      bool Looping() const         { return mPlayMode == PLAY_LOOPED; }
+      bool Scrubbing() const       { return mPlayMode == PLAY_SCRUB || mPlayMode == PLAY_KEYBOARD_SCRUB; }
+      bool PlayingAtSpeed() const  { return mPlayMode == PLAY_AT_SPEED; }
+      bool Interactive() const     { return Scrubbing() || PlayingAtSpeed(); }
+
+      // Returns true if a loop pass, or the sole pass of straight play,
+      // is completed at the current value of mTime
+      bool PassIsComplete() const;
+
+      // Returns true if time equals t1 or is on opposite side of t1, to t0
+      bool Overruns( double trackTime ) const;
+
+      // Compute the NEW track time for the given one and a real duration,
+      // taking into account whether the schedule is for looping
+      double AdvancedTrackTime(
+         double trackTime, double realElapsed, double speed) const;
+
+      // Use the function above in the callback after consuming samples from the
+      // playback ring buffers, during usual straight or looping play
+      void TrackTimeUpdate(double realElapsed);
+
+      // Convert time between mT0 and argument to real duration, according to
+      // time track if one is given; result is always nonnegative
+      double RealDuration(double trackTime1) const;
+
+      // How much real time left?
+      double RealTimeRemaining() const;
+
+      // Advance the real time position
+      void RealTimeAdvance( double increment );
+
+      // Determine starting duration within the first pass -- sometimes not
+      // zero
+      void RealTimeInit( double trackTime );
+      
+      void RealTimeRestart();
+
+   };
+
+   /** \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(const wxString &devName = {});
+
+   /** \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(const wxString &devName = {});
+
+   /** \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;
+};
+
+#endif