630 lines
13 KiB
C++
630 lines
13 KiB
C++
/* fhandler_dev_dsp: code to emulate OSS sound model /dev/dsp
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Copyright 2001 Red Hat, Inc
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Written by Andy Younger (andy@snoogie.demon.co.uk)
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This file is part of Cygwin.
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This software is a copyrighted work licensed under the terms of the
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Cygwin license. Please consult the file "CYGWIN_LICENSE" for
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details. */
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#include "winsup.h"
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#include <stdio.h>
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#include <errno.h>
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#include "cygerrno.h"
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#include "fhandler.h"
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#include <windows.h>
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#include <sys/soundcard.h>
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#include <sys/fcntl.h>
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#include <mmsystem.h>
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//------------------------------------------------------------------------
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// Simple encapsulation of the win32 audio device.
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//
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static void CALLBACK wave_callback(HWAVE hWave, UINT msg, DWORD instance,
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DWORD param1, DWORD param2);
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class Audio
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{
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public:
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enum { MAX_BLOCKS = 12, BLOCK_SIZE = 16384 };
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Audio ();
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~Audio ();
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bool open (int rate, int bits, int channels, bool bCallback = false);
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void close ();
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int getvolume ();
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void setvolume (int newVolume);
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bool write (const void *pSampleData, int nBytes);
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int blocks ();
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void callback_sampledone (void *pData);
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void setformat(int format) { formattype_ = format; }
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int numbytesoutput ();
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private:
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char *initialisebuffer ();
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void waitforcallback ();
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bool flush ();
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HWAVEOUT dev_;
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volatile int nBlocksInQue_;
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int nBytesWritten_;
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char *buffer_;
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int bufferIndex_;
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CRITICAL_SECTION lock_;
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char *freeblocks_[MAX_BLOCKS];
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int formattype_;
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char bigwavebuffer_[MAX_BLOCKS * BLOCK_SIZE];
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};
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Audio::Audio()
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{
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int size = BLOCK_SIZE + sizeof(WAVEHDR);
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InitializeCriticalSection(&lock_);
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memset(freeblocks_, 0, sizeof(freeblocks_));
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for (int i = 0; i < MAX_BLOCKS; i++)
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{
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char *pBuffer = &bigwavebuffer_[ i * size ];
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memset(pBuffer, 0, size);
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freeblocks_[i] = pBuffer;
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}
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}
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Audio::~Audio()
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{
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if (dev_)
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close();
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DeleteCriticalSection(&lock_);
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}
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bool
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Audio::open(int rate, int bits, int channels, bool bCallback = false)
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{
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WAVEFORMATEX format;
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int nDevices = waveOutGetNumDevs();
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nBytesWritten_ = 0L;
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bufferIndex_ = 0;
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buffer_ = 0L;
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debug_printf("number devices %d\n", nDevices);
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if (nDevices <= 0)
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return false;
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debug_printf("trying to map device freq %d, bits %d, "
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"channels %d, callback %d\n", rate, bits, channels,
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bCallback);
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int bytesperSample = bits / 8;
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memset(&format, 0, sizeof(format));
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format.wFormatTag = WAVE_FORMAT_PCM;
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format.wBitsPerSample = bits;
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format.nChannels = channels;
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format.nSamplesPerSec = rate;
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format.nAvgBytesPerSec = format.nSamplesPerSec * format.nChannels *
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bytesperSample;
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format.nBlockAlign = format.nChannels * bytesperSample;
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nBlocksInQue_ = 0;
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HRESULT res = waveOutOpen(&dev_, WAVE_MAPPER, &format, (DWORD)wave_callback,
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(DWORD)this, bCallback ? CALLBACK_FUNCTION : 0);
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if (res == S_OK)
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{
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debug_printf("Sucessfully opened!");
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return true;
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}
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else
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{
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debug_printf("failed to open");
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return false;
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}
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}
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void
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Audio::close()
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{
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if (dev_)
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{
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flush(); // force out last block whatever size..
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while (blocks()) // block till finished..
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waitforcallback();
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waveOutReset(dev_);
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waveOutClose(dev_);
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dev_ = 0L;
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}
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nBytesWritten_ = 0L;
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}
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int
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Audio::numbytesoutput()
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{
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return nBytesWritten_;
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}
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int
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Audio::getvolume()
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{
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DWORD volume;
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waveOutGetVolume(dev_, &volume);
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return ((volume >> 16) + (volume & 0xffff)) >> 1;
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}
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void
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Audio::setvolume(int newVolume)
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{
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waveOutSetVolume(dev_, (newVolume<<16)|newVolume);
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}
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char *
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Audio::initialisebuffer()
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{
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EnterCriticalSection(&lock_);
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WAVEHDR *pHeader = 0L;
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for (int i = 0; i < MAX_BLOCKS; i++)
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{
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char *pData = freeblocks_[i];
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if (pData)
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{
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pHeader = (WAVEHDR *)pData;
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if (pHeader->dwFlags & WHDR_DONE)
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{
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waveOutUnprepareHeader(dev_, pHeader, sizeof(WAVEHDR));
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}
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freeblocks_[i] = 0L;
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break;
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}
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}
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LeaveCriticalSection(&lock_);
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if (pHeader)
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{
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memset(pHeader, 0, sizeof(WAVEHDR));
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pHeader->dwBufferLength = BLOCK_SIZE;
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pHeader->lpData = (LPSTR)(&pHeader[1]);
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return (char *)pHeader->lpData;
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}
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return 0L;
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}
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bool
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Audio::write(const void *pSampleData, int nBytes)
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{
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// split up big blocks into smaller BLOCK_SIZE chunks
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while (nBytes > BLOCK_SIZE)
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{
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write(pSampleData, BLOCK_SIZE);
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nBytes -= BLOCK_SIZE;
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pSampleData = (void *)((char *)pSampleData + BLOCK_SIZE);
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}
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// Block till next sound is flushed
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if (blocks() == MAX_BLOCKS)
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waitforcallback();
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// Allocate new wave buffer if necessary
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if (buffer_ == 0L)
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{
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buffer_ = initialisebuffer();
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if (buffer_ == 0L)
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return false;
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}
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// Handle gathering blocks into larger buffer
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int sizeleft = BLOCK_SIZE - bufferIndex_;
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if (nBytes < sizeleft)
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{
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memcpy(&buffer_[bufferIndex_], pSampleData, nBytes);
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bufferIndex_ += nBytes;
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nBytesWritten_ += nBytes;
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return true;
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}
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// flushing when we reach our limit of BLOCK_SIZE
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memcpy(&buffer_[bufferIndex_], pSampleData, sizeleft);
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bufferIndex_ += sizeleft;
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nBytesWritten_ += sizeleft;
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flush();
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// change pointer to rest of sample, and size accordingly
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pSampleData = (void *)((char *)pSampleData + sizeleft);
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nBytes -= sizeleft;
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// if we still have some sample left over write it out
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if (nBytes)
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return write(pSampleData, nBytes);
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return true;
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}
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// return number of blocks back.
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int
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Audio::blocks()
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{
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EnterCriticalSection(&lock_);
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int ret = nBlocksInQue_;
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LeaveCriticalSection(&lock_);
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return ret;
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}
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// This is called on an interupt so use locking.. Note nBlocksInQue_ is
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// modified by it so we should wrap all references to it in locks.
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void
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Audio::callback_sampledone(void *pData)
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{
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EnterCriticalSection(&lock_);
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nBlocksInQue_--;
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for (int i = 0; i < MAX_BLOCKS; i++)
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if (!freeblocks_[i])
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{
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freeblocks_[i] = (char *)pData;
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break;
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}
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LeaveCriticalSection(&lock_);
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}
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void
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Audio::waitforcallback()
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{
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int n = blocks();
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if (!n)
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return;
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do
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{
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Sleep(250);
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}
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while (n == blocks());
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}
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bool
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Audio::flush()
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{
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if (!buffer_)
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return false;
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// Send internal buffer out to the soundcard
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WAVEHDR *pHeader = ((WAVEHDR *)buffer_) - 1;
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pHeader->dwBufferLength = bufferIndex_;
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// Quick bit of sample buffer conversion
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if (formattype_ == AFMT_S8)
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{
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unsigned char *p = ((unsigned char *)buffer_);
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for (int i = 0; i < bufferIndex_; i++)
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{
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p[i] -= 0x7f;
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}
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}
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if (waveOutPrepareHeader(dev_, pHeader, sizeof(WAVEHDR)) == S_OK &&
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waveOutWrite(dev_, pHeader, sizeof (WAVEHDR)) == S_OK)
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{
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EnterCriticalSection(&lock_);
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nBlocksInQue_++;
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LeaveCriticalSection(&lock_);
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bufferIndex_ = 0;
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buffer_ = 0L;
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return true;
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}
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else
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{
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EnterCriticalSection(&lock_);
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for (int i = 0; i < MAX_BLOCKS; i++)
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if (!freeblocks_[i])
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{
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freeblocks_[i] = (char *)pHeader;
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break;
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}
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LeaveCriticalSection(&lock_);
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}
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return false;
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}
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//------------------------------------------------------------------------
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// Call back routine
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static void CALLBACK
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wave_callback(HWAVE hWave, UINT msg, DWORD instance, DWORD param1, DWORD param2)
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{
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if (msg == WOM_DONE)
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{
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Audio *ptr = (Audio *)instance;
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ptr->callback_sampledone((void *)param1);
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}
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}
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//------------------------------------------------------------------------
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// /dev/dsp handler
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static Audio s_audio; // static instance of the Audio handler
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//------------------------------------------------------------------------
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// wav file detection..
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#pragma pack(1)
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struct wavchunk
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{
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char id[4];
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unsigned int len;
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};
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struct wavformat
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{
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unsigned short wFormatTag;
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unsigned short wChannels;
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unsigned int dwSamplesPerSec;
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unsigned int dwAvgBytesPerSec;
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unsigned short wBlockAlign;
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unsigned short wBitsPerSample;
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};
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#pragma pack()
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bool
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fhandler_dev_dsp::setupwav(const char *pData, int nBytes)
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{
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int len;
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const char *end = pData + nBytes;
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if (!(pData[0] == 'R' && pData[1] == 'I' &&
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pData[2] == 'F' && pData[3] == 'F'))
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return false;
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if (!(pData[8] == 'W' && pData[9] == 'A' &&
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pData[10] == 'V' && pData[11] == 'E'))
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return false;
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len = *(int *)&pData[4];
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pData += 12;
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while (len && pData < end)
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{
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wavchunk *pChunk = (wavchunk *)pData;
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int blklen = pChunk->len;
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if (pChunk->id[0] == 'f' && pChunk->id[1] == 'm' &&
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pChunk->id[2] == 't' && pChunk->id[3] == ' ')
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{
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wavformat *format = (wavformat *)(pChunk+1);
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if ((char *)(format+1) > end)
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return false;
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// Open up audio device with correct frequency for wav file
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//
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// FIXME: should through away all the header & not output
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// it to the soundcard.
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s_audio.close();
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if (s_audio.open(format->dwSamplesPerSec, format->wBitsPerSample,
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format->wChannels) == false)
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{
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s_audio.open(audiofreq_, audiobits_, audiochannels_);
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}
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else
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{
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audiofreq_ = format->dwSamplesPerSec;
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audiobits_ = format->wBitsPerSample;
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audiochannels_ = format->wChannels;
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}
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return true;
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}
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pData += blklen + sizeof(wavchunk);
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}
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return false;
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}
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//------------------------------------------------------------------------
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fhandler_dev_dsp::fhandler_dev_dsp (const char *name)
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: fhandler_base (FH_OSS_DSP, name)
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{
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set_cb (sizeof *this);
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}
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fhandler_dev_dsp::~fhandler_dev_dsp()
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{
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}
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int
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fhandler_dev_dsp::open (const char *path, int flags, mode_t mode = 0)
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{
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// currently we only support writing
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if ((flags & (O_WRONLY|O_RDONLY|O_RDWR)) != O_WRONLY)
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return 0;
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set_flags(flags);
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// Work out initial sample format & frequency
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if (strcmp(path, "/dev/dsp") == 0L)
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{
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// dev/dsp defaults
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audioformat_ = AFMT_S8;
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audiofreq_ = 8000;
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audiobits_ = 8;
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audiochannels_ = 1;
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}
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if (!s_audio.open(audiofreq_, audiobits_, audiochannels_))
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debug_printf("/dev/dsp: failed to open\n");
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else
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{
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set_open_status ();
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debug_printf("/dev/dsp: successfully opened\n");
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}
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return 1;
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}
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int
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fhandler_dev_dsp::write (const void *ptr, size_t len)
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{
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if (s_audio.numbytesoutput() == 0)
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{
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// check for wave file & setup frequencys properly if possible.
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setupwav((const char *)ptr, len);
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// Open audio device properly with callbacks.
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s_audio.close();
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if (!s_audio.open(audiofreq_, audiobits_, audiochannels_, true))
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return 0;
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}
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s_audio.write(ptr, len);
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return len;
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}
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int
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fhandler_dev_dsp::read (void *ptr, size_t len)
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{
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return len;
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}
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off_t
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fhandler_dev_dsp::lseek (off_t offset, int whence)
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{
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return 0;
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}
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int
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fhandler_dev_dsp::close (void)
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{
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s_audio.close();
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return 0;
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}
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int
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fhandler_dev_dsp::dup (fhandler_base * child)
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{
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fhandler_dev_dsp *fhc = (fhandler_dev_dsp *)child;
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fhc->set_flags(get_flags());
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fhc->audiochannels_ = audiochannels_;
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fhc->audiobits_ = audiobits_;
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fhc->audiofreq_ = audiofreq_;
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fhc->audioformat_ = audioformat_;
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return 0;
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}
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int
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fhandler_dev_dsp::ioctl(unsigned int cmd, void *ptr)
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{
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int *intptr = (int *)ptr;
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switch (cmd)
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{
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#define CASE(a) case a : debug_printf("/dev/dsp: ioctl %s\n", #a);
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CASE(SNDCTL_DSP_RESET)
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audioformat_ = AFMT_S8;
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audiofreq_ = 8000;
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audiobits_ = 8;
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audiochannels_ = 1;
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return 1;
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CASE(SNDCTL_DSP_GETBLKSIZE)
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*intptr = Audio::BLOCK_SIZE;
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break;
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CASE(SNDCTL_DSP_SETFMT)
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{
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int nBits = 0;
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if (*intptr == AFMT_S16_LE)
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nBits = 16;
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else if (*intptr == AFMT_U8)
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nBits = 8;
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else if (*intptr == AFMT_S8)
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nBits = 8;
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if (nBits)
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{
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s_audio.setformat(*intptr);
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s_audio.close();
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if (s_audio.open(audiofreq_, nBits, audiochannels_) == true)
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{
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audiobits_ = nBits;
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return 1;
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}
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else
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{
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s_audio.open(audiofreq_, audiobits_, audiochannels_);
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return -1;
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}
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}
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} break;
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CASE(SNDCTL_DSP_SPEED)
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s_audio.close();
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if (s_audio.open(*intptr, audiobits_, audiochannels_) == true)
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{
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audiofreq_ = *intptr;
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return 1;
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}
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else
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{
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s_audio.open(audiofreq_, audiobits_, audiochannels_);
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return -1;
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}
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break;
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CASE(SNDCTL_DSP_STEREO)
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{
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int nChannels = *intptr + 1;
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s_audio.close();
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if (s_audio.open(audiofreq_, audiobits_, nChannels) == true)
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{
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audiochannels_ = nChannels;
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return 1;
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}
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else
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{
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s_audio.open(audiofreq_, audiobits_, audiochannels_);
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return -1;
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}
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} break;
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CASE(SNDCTL_DSP_GETOSPACE)
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{
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audio_buf_info *p = (audio_buf_info *)ptr;
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int nBlocks = s_audio.blocks();
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int leftblocks = ((Audio::MAX_BLOCKS - nBlocks)-1);
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if (leftblocks < 0) leftblocks = 0;
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if (leftblocks > 1)
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leftblocks = 1;
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int left = leftblocks * Audio::BLOCK_SIZE;
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p->fragments = leftblocks;
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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");
|
|
}
|
|
|