mirror of
git://sourceware.org/git/newlib-cygwin.git
synced 2025-01-18 12:29:32 +08:00
* fhandler_dsp.cc: Reformat to GNU standards.
(s_audio): Change to a pointer throughout. (fhandler_dev_dsp::open): Initialize s_audio, if required.
This commit is contained in:
Christopher Faylor
parent
1fcc912f13
commit
1b72f36e89
@ -1,3 +1,9 @@
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Sun May 20 13:26:25 2001 Christopher Faylor <cgf@cygnus.com>
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* fhandler_dsp.cc: Reformat to GNU standards.
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(s_audio): Change to a pointer throughout.
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(fhandler_dev_dsp::open): Initialize s_audio, if required.
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Sat May 19 23:40:00 2001 Corinna Vinschen <corinna@vinschen.de>
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* autoload.cc: Add load statements for `LookupAccountNameW',
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@ -1,11 +1,11 @@
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/* 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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@ -22,17 +22,17 @@ details. */
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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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//
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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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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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@ -40,14 +40,14 @@ public:
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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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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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@ -60,296 +60,297 @@ private:
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char bigwavebuffer_[MAX_BLOCKS * BLOCK_SIZE];
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};
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Audio::Audio()
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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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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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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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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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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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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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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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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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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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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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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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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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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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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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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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{
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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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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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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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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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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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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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waveOutSetVolume (dev_, (newVolume << 16) | newVolume);
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}
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char *
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Audio::initialisebuffer()
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Audio::initialisebuffer ()
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{
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EnterCriticalSection(&lock_);
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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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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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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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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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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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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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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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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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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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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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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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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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// 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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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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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 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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int
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Audio::blocks ()
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{
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EnterCriticalSection(&lock_);
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EnterCriticalSection (&lock_);
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int ret = nBlocksInQue_;
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LeaveCriticalSection(&lock_);
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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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// 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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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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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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freeblocks_[i] = (char *) pData;
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break;
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}
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LeaveCriticalSection(&lock_);
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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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Audio::waitforcallback ()
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{
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int n = blocks();
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int n = blocks ();
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if (!n)
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return;
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return;
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do
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{
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Sleep(250);
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Sleep (250);
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}
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while (n == blocks());
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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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Audio::flush ()
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{
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if (!buffer_)
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return false;
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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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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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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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{
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p[i] -= 0x7f;
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}
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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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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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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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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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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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static void CALLBACK
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wave_callback (HWAVE hWave, UINT msg, DWORD instance, DWORD param1,
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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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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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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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{
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char id[4];
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unsigned int len;
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};
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@ -365,40 +366,40 @@ struct wavformat
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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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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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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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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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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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{
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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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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
|
||||
//
|
||||
//
|
||||
// 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->close ();
|
||||
if (s_audio->open (format->dwSamplesPerSec, format->wBitsPerSample,
|
||||
format->wChannels) == false)
|
||||
{
|
||||
s_audio.open(audiofreq_, audiobits_, audiochannels_);
|
||||
s_audio->open (audiofreq_, audiobits_, audiochannels_);
|
||||
}
|
||||
else
|
||||
{
|
||||
@ -409,33 +410,36 @@ fhandler_dev_dsp::setupwav(const char *pData, int nBytes)
|
||||
return true;
|
||||
}
|
||||
|
||||
pData += blklen + sizeof(wavchunk);
|
||||
pData += blklen + sizeof (wavchunk);
|
||||
}
|
||||
return false;
|
||||
}
|
||||
|
||||
//------------------------------------------------------------------------
|
||||
fhandler_dev_dsp::fhandler_dev_dsp (const char *name)
|
||||
: fhandler_base (FH_OSS_DSP, name)
|
||||
fhandler_dev_dsp::fhandler_dev_dsp (const char *name):
|
||||
fhandler_base (FH_OSS_DSP, name)
|
||||
{
|
||||
set_cb (sizeof *this);
|
||||
set_cb (sizeof *this);
|
||||
}
|
||||
|
||||
fhandler_dev_dsp::~fhandler_dev_dsp()
|
||||
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)
|
||||
// currently we only support writing
|
||||
if ((flags & (O_WRONLY | O_RDONLY | O_RDWR)) != O_WRONLY)
|
||||
return 0;
|
||||
|
||||
set_flags(flags);
|
||||
|
||||
set_flags (flags);
|
||||
|
||||
if (!s_audio)
|
||||
s_audio = new Audio;
|
||||
|
||||
// Work out initial sample format & frequency
|
||||
if (strcmp(path, "/dev/dsp") == 0L)
|
||||
if (strcmp (path, "/dev/dsp") == 0L)
|
||||
{
|
||||
// dev/dsp defaults
|
||||
audioformat_ = AFMT_S8;
|
||||
@ -444,12 +448,12 @@ fhandler_dev_dsp::open (const char *path, int flags, mode_t mode = 0)
|
||||
audiochannels_ = 1;
|
||||
}
|
||||
|
||||
if (!s_audio.open(audiofreq_, audiobits_, audiochannels_))
|
||||
debug_printf("/dev/dsp: failed to open\n");
|
||||
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");
|
||||
debug_printf ("/dev/dsp: successfully opened\n");
|
||||
}
|
||||
return 1;
|
||||
}
|
||||
@ -457,18 +461,18 @@ fhandler_dev_dsp::open (const char *path, int flags, mode_t mode = 0)
|
||||
int
|
||||
fhandler_dev_dsp::write (const void *ptr, size_t len)
|
||||
{
|
||||
if (s_audio.numbytesoutput() == 0)
|
||||
if (s_audio->numbytesoutput () == 0)
|
||||
{
|
||||
// check for wave file & setup frequencys properly if possible.
|
||||
setupwav((const char *)ptr, len);
|
||||
|
||||
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->close ();
|
||||
if (!s_audio->open (audiofreq_, audiobits_, audiochannels_, true))
|
||||
return 0;
|
||||
}
|
||||
|
||||
s_audio.write(ptr, len);
|
||||
|
||||
s_audio->write (ptr, len);
|
||||
return len;
|
||||
}
|
||||
|
||||
@ -487,16 +491,16 @@ fhandler_dev_dsp::lseek (off_t offset, int whence)
|
||||
int
|
||||
fhandler_dev_dsp::close (void)
|
||||
{
|
||||
s_audio.close();
|
||||
s_audio->close ();
|
||||
return 0;
|
||||
}
|
||||
|
||||
int
|
||||
fhandler_dev_dsp::dup (fhandler_base * child)
|
||||
{
|
||||
fhandler_dev_dsp *fhc = (fhandler_dev_dsp *)child;
|
||||
fhandler_dev_dsp *fhc = (fhandler_dev_dsp *) child;
|
||||
|
||||
fhc->set_flags(get_flags());
|
||||
fhc->set_flags (get_flags ());
|
||||
fhc->audiochannels_ = audiochannels_;
|
||||
fhc->audiobits_ = audiobits_;
|
||||
fhc->audiofreq_ = audiofreq_;
|
||||
@ -504,119 +508,121 @@ fhandler_dev_dsp::dup (fhandler_base * child)
|
||||
return 0;
|
||||
}
|
||||
|
||||
int
|
||||
fhandler_dev_dsp::ioctl(unsigned int cmd, void *ptr)
|
||||
int
|
||||
fhandler_dev_dsp::ioctl (unsigned int cmd, void *ptr)
|
||||
{
|
||||
int *intptr = (int *)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;
|
||||
#define CASE(a) case a : debug_printf("/dev/dsp: ioctl %s\n", #a);
|
||||
|
||||
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_RESET)
|
||||
audioformat_ = AFMT_S8;
|
||||
audiofreq_ = 8000;
|
||||
audiobits_ = 8;
|
||||
audiochannels_ = 1;
|
||||
return 1;
|
||||
|
||||
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");
|
||||
CASE (SNDCTL_DSP_GETBLKSIZE)
|
||||
*intptr = Audio::BLOCK_SIZE;
|
||||
break;
|
||||
|
||||
#undef CASE
|
||||
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;
|
||||
}
|
||||
@ -624,6 +630,5 @@ fhandler_dev_dsp::ioctl(unsigned int cmd, void *ptr)
|
||||
void
|
||||
fhandler_dev_dsp::dump ()
|
||||
{
|
||||
paranoid_printf("here, fhandler_dev_dsp");
|
||||
paranoid_printf ("here, fhandler_dev_dsp");
|
||||
}
|
||||
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user