libc/winsup/cygwin/fhandler_dsp.cc

/* fhandler_dev_dsp: code to emulate OSS sound model /dev/dsp

   Copyright 2001, 2002, 2003 Red Hat, Inc

   Written by Andy Younger (andy@snoogie.demon.co.uk)

This file is part of Cygwin.

This software is a copyrighted work licensed under the terms of the
Cygwin license.  Please consult the file "CYGWIN_LICENSE" for
details. */

#include "winsup.h"
#include <stdio.h>
#include <windows.h>
#include <sys/soundcard.h>
#include <mmsystem.h>
#include "cygerrno.h"
#include "security.h"
#include "path.h"
#include "fhandler.h"

//------------------------------------------------------------------------
// Simple encapsulation of the win32 audio device.
//
static void CALLBACK wave_callback (HWAVE hWave, UINT msg, DWORD instance,
				    DWORD param1, DWORD param2);
class Audio
{
public:
  enum
  {
    MAX_BLOCKS = 12,
    BLOCK_SIZE = 16384,
    TOT_BLOCK_SIZE = BLOCK_SIZE + sizeof (WAVEHDR)
   };

    Audio ();
   ~Audio ();

  bool open (int rate, int bits, int channels, bool bCallback = false);
  void close ();
  int getvolume ();
  void setvolume (int newVolume);
  bool write (const void *pSampleData, int nBytes);
  int blocks ();
  void callback_sampledone (void *pData);
  void setformat (int format) {formattype_ = format;}
  int numbytesoutput ();

  void *operator new (size_t, void *p) {return p;}

private:
  char *initialisebuffer ();
  void waitforcallback ();
  bool flush ();

  HWAVEOUT dev_;
  volatile int nBlocksInQue_;
  int nBytesWritten_;
  char *buffer_;
  int bufferIndex_;
  CRITICAL_SECTION lock_;
  char *freeblocks_[MAX_BLOCKS];
  int formattype_;

  char bigwavebuffer_[MAX_BLOCKS * TOT_BLOCK_SIZE];
};

static char audio_buf[sizeof (class Audio)];

Audio::Audio ()
{
  InitializeCriticalSection (&lock_);
  memset (bigwavebuffer_, 0, sizeof (bigwavebuffer_));
  for (int i = 0; i < MAX_BLOCKS; i++)
    freeblocks_[i] =  &bigwavebuffer_[i * TOT_BLOCK_SIZE];
}

Audio::~Audio ()
{
  if (dev_)
    close ();
  DeleteCriticalSection (&lock_);
}

bool
Audio::open (int rate, int bits, int channels, bool bCallback)
{
  WAVEFORMATEX format;
  int nDevices = waveOutGetNumDevs ();

  nBytesWritten_ = 0L;
  bufferIndex_ = 0;
  buffer_ = 0L;
  debug_printf ("number devices %d", nDevices);
  if (nDevices <= 0)
    return false;

  debug_printf ("trying to map device freq %d, bits %d, "
		"channels %d, callback %d", rate, bits, channels,
		bCallback);

  int bytesperSample = bits / 8;

  memset (&format, 0, sizeof (format));
  format.wFormatTag = WAVE_FORMAT_PCM;
  format.wBitsPerSample = bits;
  format.nChannels = channels;
  format.nSamplesPerSec = rate;
  format.nAvgBytesPerSec = format.nSamplesPerSec * format.nChannels *
    bytesperSample;
  format.nBlockAlign = format.nChannels * bytesperSample;

  nBlocksInQue_ = 0;
  HRESULT res = waveOutOpen (&dev_, WAVE_MAPPER, &format, (DWORD) wave_callback,
			     (DWORD) this, bCallback ? CALLBACK_FUNCTION : 0);
  if (res == S_OK)
    {
      debug_printf ("Sucessfully opened!");
      return true;
    }
  else
    {
      debug_printf ("failed to open");
      return false;
    }
}

void
Audio::close ()
{
  if (dev_)
    {
      flush ();			// force out last block whatever size..

      while (blocks ())		// block till finished..
	waitforcallback ();

      waveOutReset (dev_);
      waveOutClose (dev_);
      dev_ = 0L;
    }
  nBytesWritten_ = 0L;
}

int
Audio::numbytesoutput ()
{
  return nBytesWritten_;
}

int
Audio::getvolume ()
{
  DWORD volume;

  waveOutGetVolume (dev_, &volume);
  return ((volume >> 16) + (volume & 0xffff)) >> 1;
}

void
Audio::setvolume (int newVolume)
{
  waveOutSetVolume (dev_, (newVolume << 16) | newVolume);
}

char *
Audio::initialisebuffer ()
{
  EnterCriticalSection (&lock_);
  WAVEHDR *pHeader = 0L;
  for (int i = 0; i < MAX_BLOCKS; i++)
    {
      char *pData = freeblocks_[i];
      if (pData)
	{
	  pHeader = (WAVEHDR *) pData;
	  if (pHeader->dwFlags & WHDR_DONE)
	    {
	      waveOutUnprepareHeader (dev_, pHeader, sizeof (WAVEHDR));
	    }
	  freeblocks_[i] = 0L;
	  break;
	}
    }
  LeaveCriticalSection (&lock_);

  if (pHeader)
    {
      memset (pHeader, 0, sizeof (WAVEHDR));
      pHeader->dwBufferLength = BLOCK_SIZE;
      pHeader->lpData = (LPSTR) (&pHeader[1]);
      return (char *) pHeader->lpData;
    }
  return 0L;
}

bool
Audio::write (const void *pSampleData, int nBytes)
{
  // split up big blocks into smaller BLOCK_SIZE chunks
  while (nBytes > BLOCK_SIZE)
    {
      write (pSampleData, BLOCK_SIZE);
      nBytes -= BLOCK_SIZE;
      pSampleData = (void *) ((char *) pSampleData + BLOCK_SIZE);
    }

  // Block till next sound is flushed
  if (blocks () == MAX_BLOCKS)
    waitforcallback ();

  // Allocate new wave buffer if necessary
  if (buffer_ == 0L)
    {
      buffer_ = initialisebuffer ();
      if (buffer_ == 0L)
	return false;
    }


  // Handle gathering blocks into larger buffer
  int sizeleft = BLOCK_SIZE - bufferIndex_;
  if (nBytes < sizeleft)
    {
      memcpy (&buffer_[bufferIndex_], pSampleData, nBytes);
      bufferIndex_ += nBytes;
      nBytesWritten_ += nBytes;
      return true;
    }

  // flushing when we reach our limit of BLOCK_SIZE
  memcpy (&buffer_[bufferIndex_], pSampleData, sizeleft);
  bufferIndex_ += sizeleft;
  nBytesWritten_ += sizeleft;
  flush ();

  // change pointer to rest of sample, and size accordingly
  pSampleData = (void *) ((char *) pSampleData + sizeleft);
  nBytes -= sizeleft;

  // if we still have some sample left over write it out
  if (nBytes)
    return write (pSampleData, nBytes);

  return true;
}

// return number of blocks back.
int
Audio::blocks ()
{
  EnterCriticalSection (&lock_);
  int ret = nBlocksInQue_;
  LeaveCriticalSection (&lock_);
  return ret;
}

// This is called on an interupt so use locking.. Note nBlocksInQue_ is
// modified by it so we should wrap all references to it in locks.
void
Audio::callback_sampledone (void *pData)
{
  EnterCriticalSection (&lock_);

  nBlocksInQue_--;
  for (int i = 0; i < MAX_BLOCKS; i++)
    if (!freeblocks_[i])
      {
	freeblocks_[i] = (char *) pData;
	break;
      }

  LeaveCriticalSection (&lock_);
}

void
Audio::waitforcallback ()
{
  int n = blocks ();
  if (!n)
    return;
  do
    {
      Sleep (250);
    }
  while (n == blocks ());
}

bool
Audio::flush ()
{
  if (!buffer_)
    return false;

  // Send internal buffer out to the soundcard
  WAVEHDR *pHeader = ((WAVEHDR *) buffer_) - 1;
  pHeader->dwBufferLength = bufferIndex_;

  // Quick bit of sample buffer conversion
  if (formattype_ == AFMT_S8)
    {
      unsigned char *p = ((unsigned char *) buffer_);
      for (int i = 0; i < bufferIndex_; i++)
	{
	  p[i] -= 0x7f;
	}
    }

  if (waveOutPrepareHeader (dev_, pHeader, sizeof (WAVEHDR)) == S_OK &&
      waveOutWrite (dev_, pHeader, sizeof (WAVEHDR)) == S_OK)
    {
      EnterCriticalSection (&lock_);
      nBlocksInQue_++;
      LeaveCriticalSection (&lock_);
      bufferIndex_ = 0;
      buffer_ = 0L;
      return true;
    }
  else
    {
      EnterCriticalSection (&lock_);
      for (int i = 0; i < MAX_BLOCKS; i++)
	if (!freeblocks_[i])
	  {
	    freeblocks_[i] = (char *) pHeader;
	    break;
	  }
      LeaveCriticalSection (&lock_);
    }
  return false;
}

//------------------------------------------------------------------------
// Call back routine
static void CALLBACK
wave_callback (HWAVE hWave, UINT msg, DWORD instance, DWORD param1,
	       DWORD param2)
{
  if (msg == WOM_DONE)
    {
      Audio *ptr = (Audio *) instance;
      ptr->callback_sampledone ((void *) param1);
    }
}

//------------------------------------------------------------------------
// /dev/dsp handler
static Audio *s_audio;		// static instance of the Audio handler

//------------------------------------------------------------------------
// wav file detection..
#pragma pack(1)
struct wavchunk
{
  char id[4];
  unsigned int len;
};
struct wavformat
{
  unsigned short wFormatTag;
  unsigned short wChannels;
  unsigned int dwSamplesPerSec;
  unsigned int dwAvgBytesPerSec;
  unsigned short wBlockAlign;
  unsigned short wBitsPerSample;
};
#pragma pack()

bool
fhandler_dev_dsp::setupwav (const char *pData, int nBytes)
{
  int len;
  const char *end = pData + nBytes;

  if (!(pData[0] == 'R' && pData[1] == 'I' &&
	pData[2] == 'F' && pData[3] == 'F'))
    return false;
  if (!(pData[8] == 'W' && pData[9] == 'A' &&
	pData[10] == 'V' && pData[11] == 'E'))
    return false;

  len = *(int *) &pData[4];
  pData += 12;
  while (len && pData < end)
    {
      wavchunk * pChunk = (wavchunk *) pData;
      int blklen = pChunk-> len;
      if (pChunk->id[0] == 'f' && pChunk->id[1] == 'm' &&
	  pChunk->id[2] == 't' && pChunk->id[3] == ' ')
	{
	  wavformat *format = (wavformat *) (pChunk + 1);
	  if ((char *) (format + 1) > end)
	    return false;

	  // Open up audio device with correct frequency for wav file
	  //
	  // FIXME: should through away all the header & not output
	  // it to the soundcard.
	  s_audio->close ();
	  if (s_audio->open (format->dwSamplesPerSec, format->wBitsPerSample,
			     format->wChannels) == false)
	    {
	      s_audio->open (audiofreq_, audiobits_, audiochannels_);
	    }
	  else
	    {
	      audiofreq_ = format->dwSamplesPerSec;
	      audiobits_ = format->wBitsPerSample;
	      audiochannels_ = format->wChannels;
	    }
	  return true;
	}

      pData += blklen + sizeof (wavchunk);
    }
  return false;
}

//------------------------------------------------------------------------
fhandler_dev_dsp::fhandler_dev_dsp ():
  fhandler_base ()
{
}

fhandler_dev_dsp::~fhandler_dev_dsp ()
{
}

int
fhandler_dev_dsp::open (int flags, mode_t mode)
{
  // currently we only support writing
  if ((flags & (O_WRONLY | O_RDONLY | O_RDWR)) != O_WRONLY)
    {
      set_errno (EACCES);
      return 0;
    }

  set_flags ((flags & ~O_TEXT) | O_BINARY);

  if (!s_audio)
    s_audio = new (audio_buf) Audio;

  // Work out initial sample format & frequency
      // dev/dsp defaults
  audioformat_ = AFMT_S8;
  audiofreq_ = 8000;
  audiobits_ = 8;
  audiochannels_ = 1;

  int res;
  if (!s_audio->open (audiofreq_, audiobits_, audiochannels_))
    res = 0;
  else
    {
      set_open_status ();
      res = 1;
    }

  debug_printf ("returns %d", res);
  return res;
}

int
fhandler_dev_dsp::write (const void *ptr, size_t len)
{
  if (s_audio->numbytesoutput () == 0)
    {
      // check for wave file & setup frequencys properly if possible.
      setupwav ((const char *) ptr, len);

      // Open audio device properly with callbacks.
      s_audio->close ();
      if (!s_audio->open (audiofreq_, audiobits_, audiochannels_, true))
	return 0;
    }

  s_audio->write (ptr, len);
  return len;
}

void __stdcall
fhandler_dev_dsp::read (void *ptr, size_t& len)
{
  return;
}

_off64_t
fhandler_dev_dsp::lseek (_off64_t offset, int whence)
{
  return 0;
}

int
fhandler_dev_dsp::close (void)
{
  s_audio->close ();
  return 0;
}

int
fhandler_dev_dsp::dup (fhandler_base * child)
{
  fhandler_dev_dsp *fhc = (fhandler_dev_dsp *) child;

  fhc->set_flags (get_flags ());
  fhc->audiochannels_ = audiochannels_;
  fhc->audiobits_ = audiobits_;
  fhc->audiofreq_ = audiofreq_;
  fhc->audioformat_ = audioformat_;
  return 0;
}

int
fhandler_dev_dsp::ioctl (unsigned int cmd, void *ptr)
{
  int *intptr = (int *) ptr;
  switch (cmd)
    {
#define CASE(a) case a : debug_printf("/dev/dsp: ioctl %s", #a);

      CASE (SNDCTL_DSP_RESET)
	audioformat_ = AFMT_S8;
	audiofreq_ = 8000;
	audiobits_ = 8;
	audiochannels_ = 1;
	return 0;

      CASE (SNDCTL_DSP_GETBLKSIZE)
	*intptr = Audio::BLOCK_SIZE;
	return 0;

      CASE (SNDCTL_DSP_SETFMT)
      {
	int nBits = 0;
	if (*intptr == AFMT_S16_LE)
	  nBits = 16;
	else if (*intptr == AFMT_U8)
	  nBits = 8;
	else if (*intptr == AFMT_S8)
	  nBits = 8;
	if (nBits)
	  {
	    s_audio->setformat (*intptr);
	    s_audio->close ();
	    if (s_audio->open (audiofreq_, nBits, audiochannels_) == true)
	      {
		audiobits_ = nBits;
		return 0;
	      }
	    else
	      {
		s_audio->open (audiofreq_, audiobits_, audiochannels_);
		return -1;
	      }
	  }
      }
      break;

      CASE (SNDCTL_DSP_SPEED)
	s_audio->close ();
	if (s_audio->open (*intptr, audiobits_, audiochannels_) == true)
	  {
	    audiofreq_ = *intptr;
	    return 0;
	  }
	else
	  {
	    s_audio->open (audiofreq_, audiobits_, audiochannels_);
	    return -1;
	  }
	break;

      CASE (SNDCTL_DSP_STEREO)
      {
	int nChannels = *intptr + 1;

	s_audio->close ();
	if (s_audio->open (audiofreq_, audiobits_, nChannels) == true)
	  {
	    audiochannels_ = nChannels;
	    return 0;
	  }
	else
	  {
	    s_audio->open (audiofreq_, audiobits_, audiochannels_);
	    return -1;
	  }
      }
      break;

      CASE (SNDCTL_DSP_GETOSPACE)
      {
	audio_buf_info *p = (audio_buf_info *) ptr;

	int nBlocks = s_audio->blocks ();
	int leftblocks = ((Audio::MAX_BLOCKS - nBlocks) - 1);
	if (leftblocks < 0)
	  leftblocks = 0;
	if (leftblocks > 1)
	  leftblocks = 1;
	int left = leftblocks * Audio::BLOCK_SIZE;

	p->fragments = leftblocks;
	p->fragstotal = Audio::MAX_BLOCKS;
	p->fragsize = Audio::BLOCK_SIZE;
	p->bytes = left;

	debug_printf ("ptr %p nblocks %d leftblocks %d left bytes %d ",
		      ptr, nBlocks, leftblocks, left);

	return 0;
      }
      break;

      CASE (SNDCTL_DSP_SETFRAGMENT)
      {
	// Fake!! esound & mikmod require this on non PowerPC platforms.
	//
	return 0;
      }
      break;

      CASE (SNDCTL_DSP_GETFMTS)
      {
	*intptr = AFMT_S16_LE | AFMT_U8 | AFMT_S8; // more?
	return 0;
      }
      break;

    default:
      debug_printf ("/dev/dsp: ioctl not handled yet! FIXME:");
      break;

#undef CASE
    };
  return -1;
}

void
fhandler_dev_dsp::dump ()
{
  paranoid_printf ("here, fhandler_dev_dsp");
}

void
fhandler_dev_dsp::fixup_after_exec (HANDLE)
{
  /* FIXME:  Is there a better way to do this? */
  s_audio = new (audio_buf) Audio;
}