newlib/winsup/cygwin/fhandler_dsp.cc

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/* fhandler_dev_dsp: code to emulate OSS sound model /dev/dsp
Copyright 2001 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 <errno.h>
#include "cygerrno.h"
#include "fhandler.h"
#include <windows.h>
#include <sys/soundcard.h>
#include <sys/fcntl.h>
#include <mmsystem.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 };
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 ();
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 * BLOCK_SIZE];
};
Audio::Audio()
{
int size = BLOCK_SIZE + sizeof(WAVEHDR);
InitializeCriticalSection(&lock_);
memset(freeblocks_, 0, sizeof(freeblocks_));
for (int i = 0; i < MAX_BLOCKS; i++)
{
char *pBuffer = &bigwavebuffer_[ i * size ];
memset(pBuffer, 0, size);
freeblocks_[i] = pBuffer;
}
}
Audio::~Audio()
{
if (dev_)
close();
DeleteCriticalSection(&lock_);
}
bool
Audio::open(int rate, int bits, int channels, bool bCallback = false)
{
WAVEFORMATEX format;
int nDevices = waveOutGetNumDevs();
nBytesWritten_ = 0L;
bufferIndex_ = 0;
buffer_ = 0L;
debug_printf("number devices %d\n", nDevices);
if (nDevices <= 0)
return false;
debug_printf("trying to map device freq %d, bits %d, "
"channels %d, callback %d\n", 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 (const char *name)
: fhandler_base (FH_OSS_DSP, name)
{
set_cb (sizeof *this);
}
fhandler_dev_dsp::~fhandler_dev_dsp()
{
}
int
fhandler_dev_dsp::open (const char *path, int flags, mode_t mode = 0)
{
// currently we only support writing
if ((flags & (O_WRONLY|O_RDONLY|O_RDWR)) != O_WRONLY)
return 0;
set_flags(flags);
// Work out initial sample format & frequency
if (strcmp(path, "/dev/dsp") == 0L)
{
// dev/dsp defaults
audioformat_ = AFMT_S8;
audiofreq_ = 8000;
audiobits_ = 8;
audiochannels_ = 1;
}
if (!s_audio.open(audiofreq_, audiobits_, audiochannels_))
debug_printf("/dev/dsp: failed to open\n");
else
{
set_open_status ();
debug_printf("/dev/dsp: successfully opened\n");
}
return 1;
}
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;
}
int
fhandler_dev_dsp::read (void *ptr, size_t len)
{
return len;
}
off_t
fhandler_dev_dsp::lseek (off_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\n", #a);
CASE(SNDCTL_DSP_RESET)
audioformat_ = AFMT_S8;
audiofreq_ = 8000;
audiobits_ = 8;
audiochannels_ = 1;
return 1;
CASE(SNDCTL_DSP_GETBLKSIZE)
*intptr = Audio::BLOCK_SIZE;
break;
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 1;
}
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 1;
}
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 1;
}
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 1;
} break;
CASE(SNDCTL_DSP_SETFRAGMENT)
{
// Fake!! esound & mikmod require this on non PowerPC platforms.
//
return 1;
} break;
default:
debug_printf("/dev/dsp: ioctl not handled yet! FIXME:\n");
break;
#undef CASE
};
return -1;
}
void
fhandler_dev_dsp::dump ()
{
paranoid_printf("here, fhandler_dev_dsp");
}