diff options
Diffstat (limited to 'winsup/cygwin/fhandler_dsp.cc')
| -rw-r--r-- | winsup/cygwin/fhandler_dsp.cc | 1372 |
1 files changed, 0 insertions, 1372 deletions
diff --git a/winsup/cygwin/fhandler_dsp.cc b/winsup/cygwin/fhandler_dsp.cc deleted file mode 100644 index ed5ddad8a..000000000 --- a/winsup/cygwin/fhandler_dsp.cc +++ /dev/null @@ -1,1372 +0,0 @@ -/* Fhandler_dev_dsp: code to emulate OSS sound model /dev/dsp - - Copyright 2001, 2002, 2003, 2004, 2008 Red Hat, Inc - - Written by Andy Younger (andy@snoogie.demon.co.uk) - Extended by Gerd Spalink (Gerd.Spalink@t-online.de) - to support recording from the audio input - -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 <sys/soundcard.h> -#include "cygerrno.h" -#include "security.h" -#include "path.h" -#include "fhandler.h" -#include "dtable.h" -#include "cygheap.h" - -/*------------------------------------------------------------------------ - Simple encapsulation of the win32 audio device. - - Implementation Notes - 1. Audio structures are malloced just before the first read or - write to /dev/dsp. The actual buffer size is determined at that time, - such that one buffer holds about 125ms of audio data. - At the time of this writing, 12 buffers are allocated, - so that up to 1.5 seconds can be buffered within Win32. - The buffer size can be queried with the ioctl SNDCTL_DSP_GETBLKSIZE, - but for this implementation only returns meaningful results if - sampling rate, number of channels and number of bits per sample - are not changed afterwards. - The audio structures are freed when the device is reset or closed, - and they are not passed to exec'ed processes. - The dev_ member is cleared after a fork. This forces the child - to reopen the audio device._ - - 2. Every open call creates a new instance of the handler. After a - successful open, every subsequent open from the same process - to the device fails with EBUSY. - The structures are shared between duped handles, but not with - children. They only inherit the settings from the parent. - */ - -class fhandler_dev_dsp::Audio -{ // This class contains functionality common to Audio_in and Audio_out - public: - Audio (); - ~Audio (); - - class queue; - - bool isvalid (); - void setconvert (int format); - void convert_none (unsigned char *buffer, int size_bytes) { } - void convert_U8_S8 (unsigned char *buffer, int size_bytes); - void convert_S16LE_U16LE (unsigned char *buffer, int size_bytes); - void convert_S16LE_U16BE (unsigned char *buffer, int size_bytes); - void convert_S16LE_S16BE (unsigned char *buffer, int size_bytes); - void fillFormat (WAVEFORMATEX * format, - int rate, int bits, int channels); - unsigned blockSize (int rate, int bits, int channels); - void (fhandler_dev_dsp::Audio::*convert_) - (unsigned char *buffer, int size_bytes); - - enum { MAX_BLOCKS = 12 }; - int bufferIndex_; // offset into pHdr_->lpData - WAVEHDR *pHdr_; // data to be filled by write - WAVEHDR wavehdr_[MAX_BLOCKS]; - char *bigwavebuffer_; // audio samples only - // Member variables below must be locked - queue *Qisr2app_; // blocks passed from wave callback -}; - -class fhandler_dev_dsp::Audio::queue -{ // non-blocking fixed size queues for buffer management - public: - queue (int depth = 4); - ~queue (); - - bool send (WAVEHDR *); // queue an item, returns true if successful - bool recv (WAVEHDR **); // retrieve an item, returns true if successful - void reset (); - int query (); // return number of items queued - inline void lock () { EnterCriticalSection (&lock_); } - inline void unlock () { LeaveCriticalSection (&lock_); } - inline void dellock () { debug_printf ("Deleting Critical Section"); DeleteCriticalSection (&lock_); } - bool isvalid () { return storage_; } - private: - CRITICAL_SECTION lock_; - int head_; - int tail_; - int depth_; - WAVEHDR **storage_; -}; - -static void CALLBACK waveOut_callback (HWAVEOUT hWave, UINT msg, DWORD instance, - DWORD param1, DWORD param2); - -class fhandler_dev_dsp::Audio_out: public Audio -{ - public: - void fork_fixup (HANDLE parent); - bool query (int rate, int bits, int channels); - bool start (); - void stop (bool immediately = false); - bool write (const char *pSampleData, int nBytes); - void buf_info (audio_buf_info *p, int rate, int bits, int channels); - void callback_sampledone (WAVEHDR *pHdr); - bool parsewav (const char *&pData, int &nBytes, - int rate, int bits, int channels); - - private: - void init (unsigned blockSize); - void waitforallsent (); - void waitforspace (); - bool sendcurrent (); - - enum { MAX_BLOCKS = 12 }; - HWAVEOUT dev_; // The wave device - /* Private copies of audiofreq_, audiobits_, audiochannels_, - possibly set from wave file */ - int freq_; - int bits_; - int channels_; -}; - -static void CALLBACK waveIn_callback (HWAVEIN hWave, UINT msg, DWORD instance, - DWORD param1, DWORD param2); - -class fhandler_dev_dsp::Audio_in: public Audio -{ -public: - void fork_fixup (HANDLE parent); - bool query (int rate, int bits, int channels); - bool start (int rate, int bits, int channels); - void stop (); - bool read (char *pSampleData, int &nBytes); - void buf_info (audio_buf_info *p, int rate, int bits, int channels); - void callback_blockfull (WAVEHDR *pHdr); - -private: - bool init (unsigned blockSize); - bool queueblock (WAVEHDR *pHdr); - void waitfordata (); // blocks until we have a good pHdr_ - - HWAVEIN dev_; -}; - -/* -------------------------------------------------------------------- - Implementation */ - -// Simple fixed length FIFO queue implementation for audio buffer management -fhandler_dev_dsp::Audio::queue::queue (int depth) -{ - // allow space for one extra object in the queue - // so we can distinguish full and empty status - depth_ = depth; - storage_ = new WAVEHDR *[depth_ + 1]; -} - -fhandler_dev_dsp::Audio::queue::~queue () -{ - delete[] storage_; -} - -void -fhandler_dev_dsp::Audio::queue::reset () - { - /* When starting, after reset and after fork */ - head_ = tail_ = 0; - debug_printf ("InitializeCriticalSection"); - memset (&lock_, 0, sizeof (lock_)); - InitializeCriticalSection (&lock_); - } - -bool -fhandler_dev_dsp::Audio::queue::send (WAVEHDR *x) -{ - bool res = false; - lock (); - if (query () == depth_) - system_printf ("Queue overflow"); - else - { - storage_[tail_] = x; - if (++tail_ > depth_) - tail_ = 0; - res = true; - } - unlock (); - return res; -} - -bool -fhandler_dev_dsp::Audio::queue::recv (WAVEHDR **x) -{ - bool res = false; - lock (); - if (query () != 0) - { - *x = storage_[head_]; - if (++head_ > depth_) - head_ = 0; - res = true; - } - unlock (); - return res; -} - -int -fhandler_dev_dsp::Audio::queue::query () -{ - int n = tail_ - head_; - if (n < 0) - n += depth_ + 1; - return n; -} - -// Audio class implements functionality need for both read and write -fhandler_dev_dsp::Audio::Audio () -{ - bigwavebuffer_ = NULL; - Qisr2app_ = new queue (MAX_BLOCKS); - convert_ = &fhandler_dev_dsp::Audio::convert_none; -} - -fhandler_dev_dsp::Audio::~Audio () -{ - debug_printf(""); - delete Qisr2app_; - delete[] bigwavebuffer_; -} - -inline bool -fhandler_dev_dsp::Audio::isvalid () -{ - return bigwavebuffer_ && Qisr2app_ && Qisr2app_->isvalid (); -} - -void -fhandler_dev_dsp::Audio::setconvert (int format) -{ - switch (format) - { - case AFMT_S8: - convert_ = &fhandler_dev_dsp::Audio::convert_U8_S8; - debug_printf ("U8_S8"); - break; - case AFMT_U16_LE: - convert_ = &fhandler_dev_dsp::Audio::convert_S16LE_U16LE; - debug_printf ("S16LE_U16LE"); - break; - case AFMT_U16_BE: - convert_ = &fhandler_dev_dsp::Audio::convert_S16LE_U16BE; - debug_printf ("S16LE_U16BE"); - break; - case AFMT_S16_BE: - convert_ = &fhandler_dev_dsp::Audio::convert_S16LE_S16BE; - debug_printf ("S16LE_S16BE"); - break; - default: - convert_ = &fhandler_dev_dsp::Audio::convert_none; - debug_printf ("none"); - } -} - -void -fhandler_dev_dsp::Audio::convert_U8_S8 (unsigned char *buffer, - int size_bytes) -{ - while (size_bytes-- > 0) - { - *buffer ^= (unsigned char)0x80; - buffer++; - } -} - -void -fhandler_dev_dsp::Audio::convert_S16LE_U16BE (unsigned char *buffer, - int size_bytes) -{ - int size_samples = size_bytes / 2; - unsigned char hi, lo; - while (size_samples-- > 0) - { - hi = buffer[0]; - lo = buffer[1]; - *buffer++ = lo; - *buffer++ = hi ^ (unsigned char)0x80; - } -} - -void -fhandler_dev_dsp::Audio::convert_S16LE_U16LE (unsigned char *buffer, - int size_bytes) -{ - int size_samples = size_bytes / 2; - while (size_samples-- > 0) - { - buffer++; - *buffer ^= (unsigned char)0x80; - buffer++; - } -} - -void -fhandler_dev_dsp::Audio::convert_S16LE_S16BE (unsigned char *buffer, - int size_bytes) -{ - int size_samples = size_bytes / 2; - unsigned char hi, lo; - while (size_samples-- > 0) - { - hi = buffer[0]; - lo = buffer[1]; - *buffer++ = lo; - *buffer++ = hi; - } -} - -void -fhandler_dev_dsp::Audio::fillFormat (WAVEFORMATEX * format, - int rate, int bits, int channels) -{ - 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 - * (bits / 8); - format->nBlockAlign = format->nChannels * (bits / 8); -} - -// calculate a good block size -unsigned -fhandler_dev_dsp::Audio::blockSize (int rate, int bits, int channels) -{ - unsigned blockSize; - blockSize = ((bits / 8) * channels * rate) / 8; // approx 125ms per block - // round up to multiple of 64 - blockSize += 0x3f; - blockSize &= ~0x3f; - return blockSize; -} - -//======================================================================= -void -fhandler_dev_dsp::Audio_out::fork_fixup (HANDLE parent) -{ - /* Null dev_. - It will be necessary to reset the queue, open the device - and create a lock when writing */ - debug_printf ("parent=0x%08x", parent); - dev_ = NULL; -} - - -bool -fhandler_dev_dsp::Audio_out::query (int rate, int bits, int channels) -{ - WAVEFORMATEX format; - MMRESULT rc; - - fillFormat (&format, rate, bits, channels); - rc = waveOutOpen (NULL, WAVE_MAPPER, &format, 0L, 0L, WAVE_FORMAT_QUERY); - debug_printf ("%d = waveOutOpen (freq=%d bits=%d channels=%d)", rc, rate, bits, channels); - return (rc == MMSYSERR_NOERROR); -} - -bool -fhandler_dev_dsp::Audio_out::start () -{ - WAVEFORMATEX format; - MMRESULT rc; - unsigned bSize = blockSize (freq_, bits_, channels_); - - if (dev_) - return true; - - /* In case of fork bigwavebuffer may already exist */ - if (!bigwavebuffer_) - bigwavebuffer_ = new char[MAX_BLOCKS * bSize]; - - if (!isvalid ()) - return false; - - fillFormat (&format, freq_, bits_, channels_); - rc = waveOutOpen (&dev_, WAVE_MAPPER, &format, (DWORD) waveOut_callback, - (DWORD) this, CALLBACK_FUNCTION); - if (rc == MMSYSERR_NOERROR) - init (bSize); - - debug_printf ("%d = waveOutOpen (freq=%d bits=%d channels=%d)", rc, freq_, bits_, channels_); - - return (rc == MMSYSERR_NOERROR); -} - -void -fhandler_dev_dsp::Audio_out::stop (bool immediately) -{ - MMRESULT rc; - WAVEHDR *pHdr; - - debug_printf ("dev_=%08x", (int)dev_); - if (dev_) - { - if (!immediately) - { - sendcurrent (); // force out last block whatever size.. - waitforallsent (); // block till finished.. - } - - rc = waveOutReset (dev_); - debug_printf ("%d = waveOutReset ()", rc); - while (Qisr2app_->recv (&pHdr)) - { - rc = waveOutUnprepareHeader (dev_, pHdr, sizeof (WAVEHDR)); - debug_printf ("%d = waveOutUnprepareHeader (0x%08x)", rc, pHdr); - } - - rc = waveOutClose (dev_); - debug_printf ("%d = waveOutClose ()", rc); - - Qisr2app_->dellock (); - } -} - -void -fhandler_dev_dsp::Audio_out::init (unsigned blockSize) -{ - int i; - - // internally queue all of our buffer for later use by write - Qisr2app_->reset (); - for (i = 0; i < MAX_BLOCKS; i++) - { - wavehdr_[i].lpData = &bigwavebuffer_[i * blockSize]; - wavehdr_[i].dwUser = (int) blockSize; - wavehdr_[i].dwFlags = 0; - if (!Qisr2app_->send (&wavehdr_[i])) - { - system_printf ("Internal Error i=%d", i); - break; // should not happen - } - } - pHdr_ = NULL; -} - -bool -fhandler_dev_dsp::Audio_out::write (const char *pSampleData, int nBytes) -{ - while (nBytes != 0) - { // Block if all blocks used until at least one is free - waitforspace (); - - int sizeleft = (int)pHdr_->dwUser - bufferIndex_; - if (nBytes < sizeleft) - { // all data fits into the current block, with some space left - memcpy (&pHdr_->lpData[bufferIndex_], pSampleData, nBytes); - bufferIndex_ += nBytes; - break; - } - else - { // data will fill up the current block - memcpy (&pHdr_->lpData[bufferIndex_], pSampleData, sizeleft); - bufferIndex_ += sizeleft; - sendcurrent (); - pSampleData += sizeleft; - nBytes -= sizeleft; - } - } - return true; -} - -void -fhandler_dev_dsp::Audio_out::buf_info (audio_buf_info *p, - int rate, int bits, int channels) -{ - p->fragstotal = MAX_BLOCKS; - if (this && dev_) - { - /* If the device is running we use the internal values, - possibly set from the wave file. */ - p->fragsize = blockSize (freq_, bits_, channels_); - p->fragments = Qisr2app_->query (); - if (pHdr_ != NULL) - p->bytes = (int)pHdr_->dwUser - bufferIndex_ - + p->fragsize * p->fragments; - else - p->bytes = p->fragsize * p->fragments; - } - else - { - p->fragsize = blockSize (rate, bits, channels); - p->fragments = MAX_BLOCKS; - p->bytes = p->fragsize * p->fragments; - } -} - -/* This is called on an interupt so use locking.. Note Qisr2app_ - is used so we should wrap all references to it in locks. */ -inline void -fhandler_dev_dsp::Audio_out::callback_sampledone (WAVEHDR *pHdr) -{ - Qisr2app_->send (pHdr); -} - -void -fhandler_dev_dsp::Audio_out::waitforspace () -{ - WAVEHDR *pHdr; - MMRESULT rc = WAVERR_STILLPLAYING; - - if (pHdr_ != NULL) - return; - while (!Qisr2app_->recv (&pHdr)) - { - debug_printf ("100ms"); - Sleep (100); - } - if (pHdr->dwFlags) - { - /* Errors are ignored here. They will probbaly cause a failure - in the subsequent PrepareHeader */ - rc = waveOutUnprepareHeader (dev_, pHdr, sizeof (WAVEHDR)); - debug_printf ("%d = waveOutUnprepareHeader (0x%08x)", rc, pHdr); - } - pHdr_ = pHdr; - bufferIndex_ = 0; -} - -void -fhandler_dev_dsp::Audio_out::waitforallsent () -{ - while (Qisr2app_->query () != MAX_BLOCKS) - { - debug_printf ("%d blocks in Qisr2app", Qisr2app_->query ()); - Sleep (100); - } -} - -// send the block described by pHdr_ and bufferIndex_ to wave device -bool -fhandler_dev_dsp::Audio_out::sendcurrent () -{ - WAVEHDR *pHdr = pHdr_; - MMRESULT rc; - debug_printf ("pHdr=0x%08x bytes=%d", pHdr, bufferIndex_); - - if (pHdr_ == NULL) - return false; - pHdr_ = NULL; - - // Sample buffer conversion - (this->*convert_) ((unsigned char *)pHdr->lpData, bufferIndex_); - - // Send internal buffer out to the soundcard - pHdr->dwBufferLength = bufferIndex_; - rc = waveOutPrepareHeader (dev_, pHdr, sizeof (WAVEHDR)); - debug_printf ("%d = waveOutPrepareHeader (0x%08x)", rc, pHdr); - if (rc == MMSYSERR_NOERROR) - { - rc = waveOutWrite (dev_, pHdr, sizeof (WAVEHDR)); - debug_printf ("%d = waveOutWrite (0x%08x)", rc, pHdr); - } - if (rc == MMSYSERR_NOERROR) - return true; - - /* FIXME: Should we return an error instead ?*/ - pHdr->dwFlags = 0; /* avoid calling UnprepareHeader again */ - Qisr2app_->send (pHdr); - return false; -} - -//------------------------------------------------------------------------ -// Call back routine -static void CALLBACK -waveOut_callback (HWAVEOUT hWave, UINT msg, DWORD instance, DWORD param1, - DWORD param2) -{ - if (msg == WOM_DONE) - { - fhandler_dev_dsp::Audio_out *ptr = - (fhandler_dev_dsp::Audio_out *) instance; - ptr->callback_sampledone ((WAVEHDR *) param1); - } -} - -//------------------------------------------------------------------------ -// 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::Audio_out::parsewav (const char * &pData, int &nBytes, - int dev_freq, int dev_bits, int dev_channels) -{ - int len; - const char *end = pData + nBytes; - const char *pDat; - int skip = 0; - - /* Start with default values from the device handler */ - freq_ = dev_freq; - bits_ = dev_bits; - channels_ = dev_channels; - setconvert (bits_ == 8 ? AFMT_U8 : AFMT_S16_LE); - - // Check alignment first: A lot of the code below depends on it - if (((int)pData & 0x3) != 0) - return false; - 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]; - len -= 12; - pDat = pData + 12; - skip = 12; - while ((len > 0) && (pDat + sizeof (wavchunk) < end)) - { /* We recognize two kinds of wavchunk: - "fmt " for the PCM parameters (only PCM supported here) - "data" for the start of PCM data */ - wavchunk * pChunk = (wavchunk *) pDat; - 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; - // We have found the parameter chunk - if (format->wFormatTag == 0x0001) - { // Micr*s*ft PCM; check if parameters work with our device - if (query (format->dwSamplesPerSec, format->wBitsPerSample, - format->wChannels)) - { // return the parameters we found - freq_ = format->dwSamplesPerSec; - bits_ = format->wBitsPerSample; - channels_ = format->wChannels; - } - } - } - else - { - if (pChunk->id[0] == 'd' && pChunk->id[1] == 'a' - && pChunk->id[2] == 't' && pChunk->id[3] == 'a') - { // throw away all the header & not output it to the soundcard. - skip += sizeof (wavchunk); - debug_printf ("Discard %d bytes wave header", skip); - pData += skip; - nBytes -= skip; - setconvert (bits_ == 8 ? AFMT_U8 : AFMT_S16_LE); - return true; - } - } - pDat += blklen + sizeof (wavchunk); - skip += blklen + sizeof (wavchunk); - len -= blklen + sizeof (wavchunk); - } - return false; -} - -/* ======================================================================== - Buffering concept for Audio_in: - On the first read, we queue all blocks of our bigwavebuffer - for reception and start the wave-in device. - We manage queues of pointers to WAVEHDR - When a block has been filled, the callback puts the corresponding - WAVEHDR pointer into a queue. - The function read() blocks (polled, sigh) until at least one good buffer - has arrived, then the data is copied into the buffer provided to read(). - After a buffer has been fully used by read(), it is queued again - to the wave-in device immediately. - The function read() iterates until all data requested has been - received, there is no way to interrupt it */ - -void -fhandler_dev_dsp::Audio_in::fork_fixup (HANDLE parent) -{ - /* Null dev_. - It will be necessary to reset the queue, open the device - and create a lock when reading */ - debug_printf ("parent=0x%08x", parent); - dev_ = NULL; -} - -bool -fhandler_dev_dsp::Audio_in::query (int rate, int bits, int channels) -{ - WAVEFORMATEX format; - MMRESULT rc; - - fillFormat (&format, rate, bits, channels); - rc = waveInOpen (NULL, WAVE_MAPPER, &format, 0L, 0L, WAVE_FORMAT_QUERY); - debug_printf ("%d = waveInOpen (freq=%d bits=%d channels=%d)", rc, rate, bits, channels); - return (rc == MMSYSERR_NOERROR); -} - -bool -fhandler_dev_dsp::Audio_in::start (int rate, int bits, int channels) -{ - WAVEFORMATEX format; - MMRESULT rc; - unsigned bSize = blockSize (rate, bits, channels); - - if (dev_) - return true; - - /* In case of fork bigwavebuffer may already exist */ - if (!bigwavebuffer_) - bigwavebuffer_ = new char[MAX_BLOCKS * bSize]; - - if (!isvalid ()) - return false; - - fillFormat (&format, rate, bits, channels); - rc = waveInOpen (&dev_, WAVE_MAPPER, &format, (DWORD) waveIn_callback, - (DWORD) this, CALLBACK_FUNCTION); - debug_printf ("%d = waveInOpen (rate=%d bits=%d channels=%d)", rc, rate, bits, channels); - - if (rc == MMSYSERR_NOERROR) - { - if (!init (bSize)) - return false; - } - return (rc == MMSYSERR_NOERROR); -} - -void -fhandler_dev_dsp::Audio_in::stop () -{ - MMRESULT rc; - WAVEHDR *pHdr; - - debug_printf ("dev_=%08x", (int)dev_); - if (dev_) - { - /* Note that waveInReset calls our callback for all incomplete buffers. - Since all the win32 wave functions appear to use a common lock, - we must not call into the wave API from the callback. - Otherwise we end up in a deadlock. */ - rc = waveInReset (dev_); - debug_printf ("%d = waveInReset ()", rc); - - while (Qisr2app_->recv (&pHdr)) - { - rc = waveInUnprepareHeader (dev_, pHdr, sizeof (WAVEHDR)); - debug_printf ("%d = waveInUnprepareHeader (0x%08x)", rc, pHdr); - } - - rc = waveInClose (dev_); - debug_printf ("%d = waveInClose ()", rc); - - Qisr2app_->dellock (); - } -} - -bool -fhandler_dev_dsp::Audio_in::queueblock (WAVEHDR *pHdr) -{ - MMRESULT rc; - rc = waveInPrepareHeader (dev_, pHdr, sizeof (WAVEHDR)); - debug_printf ("%d = waveInPrepareHeader (0x%08x)", rc, pHdr); - if (rc == MMSYSERR_NOERROR) - { - rc = waveInAddBuffer (dev_, pHdr, sizeof (WAVEHDR)); - debug_printf ("%d = waveInAddBuffer (0x%08x)", rc, pHdr); - } - if (rc == MMSYSERR_NOERROR) - return true; - - /* FIXME: Should the calling function return an error instead ?*/ - pHdr->dwFlags = 0; /* avoid calling UnprepareHeader again */ - pHdr->dwBytesRecorded = 0; /* no data will have been read */ - Qisr2app_->send (pHdr); - return false; -} - -bool -fhandler_dev_dsp::Audio_in::init (unsigned blockSize) -{ - MMRESULT rc; - int i; - - // try to queue all of our buffer for reception - Qisr2app_->reset (); - for (i = 0; i < MAX_BLOCKS; i++) - { - wavehdr_[i].lpData = &bigwavebuffer_[i * blockSize]; - wavehdr_[i].dwBufferLength = blockSize; - wavehdr_[i].dwFlags = 0; - if (!queueblock (&wavehdr_[i])) - break; - } - pHdr_ = NULL; - rc = waveInStart (dev_); - debug_printf ("%d = waveInStart (), queued=%d", rc, i); - return (rc == MMSYSERR_NOERROR); -} - -bool -fhandler_dev_dsp::Audio_in::read (char *pSampleData, int &nBytes) -{ - int bytes_to_read = nBytes; - nBytes = 0; - debug_printf ("pSampleData=%08x nBytes=%d", pSampleData, bytes_to_read); - while (bytes_to_read != 0) - { // Block till next sound has been read - waitfordata (); - - // Handle gathering our blocks into smaller or larger buffer - int sizeleft = pHdr_->dwBytesRecorded - bufferIndex_; - if (bytes_to_read < sizeleft) - { // The current buffer holds more data than requested - memcpy (pSampleData, &pHdr_->lpData[bufferIndex_], bytes_to_read); - (this->*convert_) ((unsigned char *)pSampleData, bytes_to_read); - nBytes += bytes_to_read; - bufferIndex_ += bytes_to_read; - debug_printf ("got %d", bytes_to_read); - break; // done; use remaining data in next call to read - } - else - { // not enough or exact amount in the current buffer - if (sizeleft) - { // use up what we have - memcpy (pSampleData, &pHdr_->lpData[bufferIndex_], sizeleft); - (this->*convert_) ((unsigned char *)pSampleData, sizeleft); - nBytes += sizeleft; - bytes_to_read -= sizeleft; - pSampleData += sizeleft; - debug_printf ("got %d", sizeleft); - } - queueblock (pHdr_); // re-queue this block to ISR - pHdr_ = NULL; // need to wait for a new block - // if more samples are needed, we need a new block now - } - } - debug_printf ("end nBytes=%d", nBytes); - return true; -} - -void -fhandler_dev_dsp::Audio_in::waitfordata () -{ - WAVEHDR *pHdr; - MMRESULT rc; - - if (pHdr_ != NULL) - return; - while (!Qisr2app_->recv (&pHdr)) - { - debug_printf ("100ms"); - Sleep (100); - } - if (pHdr->dwFlags) /* Zero if queued following error in queueblock */ - { - /* Errors are ignored here. They will probbaly cause a failure - in the subsequent PrepareHeader */ - rc = waveInUnprepareHeader (dev_, pHdr, sizeof (WAVEHDR)); - debug_printf ("%d = waveInUnprepareHeader (0x%08x)", rc, pHdr); - } - pHdr_ = pHdr; - bufferIndex_ = 0; -} - -void -fhandler_dev_dsp::Audio_in::buf_info (audio_buf_info *p, - int rate, int bits, int channels) -{ - p->fragstotal = MAX_BLOCKS; - p->fragsize = blockSize (rate, bits, channels); - if (this && dev_) - { - p->fragments = Qisr2app_->query (); - if (pHdr_ != NULL) - p->bytes = pHdr_->dwBytesRecorded - bufferIndex_ - + p->fragsize * p->fragments; - else - p->bytes = p->fragsize * p->fragments; - } - else - { - p->fragments = 0; - p->bytes = 0; - } -} - -inline void -fhandler_dev_dsp::Audio_in::callback_blockfull (WAVEHDR *pHdr) -{ - Qisr2app_->send (pHdr); -} - -static void CALLBACK -waveIn_callback (HWAVEIN hWave, UINT msg, DWORD instance, DWORD param1, - DWORD param2) -{ - if (msg == WIM_DATA) - { - fhandler_dev_dsp::Audio_in *ptr = - (fhandler_dev_dsp::Audio_in *) instance; - ptr->callback_blockfull ((WAVEHDR *) param1); - } -} - - -/* ------------------------------------------------------------------------ - /dev/dsp handler - ------------------------------------------------------------------------ */ -fhandler_dev_dsp::fhandler_dev_dsp (): - fhandler_base () -{ - debug_printf ("0x%08x", (int)this); - audio_in_ = NULL; - audio_out_ = NULL; -} - -int -fhandler_dev_dsp::open (int flags, mode_t mode) -{ - if (cygheap->fdtab.find_archetype (dev ())) - { - set_errno (EBUSY); - return 0; - } - int err = 0; - UINT num_in = 0, num_out = 0; - set_flags ((flags & ~O_TEXT) | O_BINARY); - // Work out initial sample format & frequency, /dev/dsp defaults - audioformat_ = AFMT_U8; - audiofreq_ = 8000; - audiobits_ = 8; - audiochannels_ = 1; - switch (flags & O_ACCMODE) - { - case O_RDWR: - if ((num_in = waveInGetNumDevs ()) == 0) - err = ENXIO; - /* Fall through */ - case O_WRONLY: - if ((num_out = waveOutGetNumDevs ()) == 0) - err = ENXIO; - break; - case O_RDONLY: - if ((num_in = waveInGetNumDevs ()) == 0) - err = ENXIO; - break; - default: - err = EINVAL; - } - - if (!err) - { - set_open_status (); - need_fork_fixup (true); - nohandle (true); - - // FIXME: Do this better someday - fhandler_dev_dsp *arch = (fhandler_dev_dsp *) cmalloc_abort (HEAP_ARCHETYPES, sizeof (*this)); - archetype = arch; - *((fhandler_dev_dsp **) cygheap->fdtab.add_archetype ()) = arch; - *arch = *this; - archetype->usecount = 1; - } - else - set_errno (err); - - debug_printf ("ACCMODE=0x%08x audio_in=%d audio_out=%d, err=%d", - flags & O_ACCMODE, num_in, num_out, err); - return !err; -} - -#define IS_WRITE() ((get_flags() & O_ACCMODE) != O_RDONLY) -#define IS_READ() ((get_flags() & O_ACCMODE) != O_WRONLY) - -int -fhandler_dev_dsp::write (const void *ptr, size_t len) -{ - debug_printf ("ptr=%08x len=%d", ptr, len); - if ((fhandler_dev_dsp *) archetype != this) - return ((fhandler_dev_dsp *)archetype)->write(ptr, len); - - int len_s = len; - const char *ptr_s = static_cast <const char *> (ptr); - - if (!audio_out_) - if (IS_WRITE ()) - { - debug_printf ("Allocating"); - if (!(audio_out_ = new Audio_out)) - return -1; - - /* check for wave file & get parameters & skip header if possible. */ - - if (audio_out_->parsewav (ptr_s, len_s, - audiofreq_, audiobits_, audiochannels_)) - debug_printf ("=> ptr_s=%08x len_s=%d", ptr_s, len_s); - } - else - { - set_errno (EBADF); // device was opened for read? - return -1; - } - - /* Open audio device properly with callbacks. - Private parameters were set in call to parsewav. - This is a no-op when there are successive writes in the same process */ - if (!audio_out_->start ()) - { - set_errno (EIO); - return -1; - } - - audio_out_->write (ptr_s, len_s); - return len; -} - -void __stdcall -fhandler_dev_dsp::read (void *ptr, size_t& len) -{ - debug_printf ("ptr=%08x len=%d", ptr, len); - if ((fhandler_dev_dsp *) archetype != this) - return ((fhandler_dev_dsp *)archetype)->read(ptr, len); - - if (!audio_in_) - if (IS_READ ()) - { - debug_printf ("Allocating"); - if (!(audio_in_ = new Audio_in)) - { - len = (size_t)-1; - return; - } - audio_in_->setconvert (audioformat_); - } - else - { - len = (size_t)-1; - set_errno (EBADF); // device was opened for write? - return; - } - - /* Open audio device properly with callbacks. - This is a noop when there are successive reads in the same process */ - if (!audio_in_->start (audiofreq_, audiobits_, audiochannels_)) - { - len = (size_t)-1; - set_errno (EIO); - return; - } - - audio_in_->read ((char *)ptr, (int&)len); -} - -_off64_t -fhandler_dev_dsp::lseek (_off64_t offset, int whence) -{ - return 0; -} - -void -fhandler_dev_dsp::close_audio_in () -{ - if (audio_in_) - { - audio_in_->stop (); - delete audio_in_; - audio_in_ = NULL; - } -} - -void -fhandler_dev_dsp::close_audio_out (bool immediately) -{ - if (audio_out_) - { - audio_out_->stop (immediately); - delete audio_out_; - audio_out_ = NULL; - } -} - -int -fhandler_dev_dsp::close () -{ - debug_printf ("audio_in=%08x audio_out=%08x", - (int)audio_in_, (int)audio_out_); - if (!hExeced) - { - if ((fhandler_dev_dsp *) archetype != this) - return ((fhandler_dev_dsp *) archetype)->close (); - - if (--usecount == 0) - { - close_audio_in (); - close_audio_out (exit_state != ES_NOT_EXITING); - } - } - return 0; -} - -int -fhandler_dev_dsp::dup (fhandler_base * child) -{ - debug_printf (""); - child->archetype = archetype; - child->set_flags (get_flags ()); - archetype->usecount++; - return 0; -} - -int -fhandler_dev_dsp::ioctl (unsigned int cmd, void *ptr) -{ - debug_printf ("audio_in=%08x audio_out=%08x", - (int)audio_in_, (int)audio_out_); - if ((fhandler_dev_dsp *) archetype != this) - return ((fhandler_dev_dsp *)archetype)->ioctl(cmd, ptr); - - int *intptr = (int *) ptr; - switch (cmd) - { -#define CASE(a) case a : debug_printf ("/dev/dsp: ioctl %s", #a); - - CASE (SNDCTL_DSP_RESET) - close_audio_in (); - close_audio_out (true); - return 0; - break; - - CASE (SNDCTL_DSP_GETBLKSIZE) - /* This is valid even if audio_X is NULL */ - if (IS_WRITE ()) - { - *intptr = audio_out_->blockSize (audiofreq_, - audiobits_, - audiochannels_); - } - else - { // I am very sure that IS_READ is valid - *intptr = audio_in_->blockSize (audiofreq_, - audiobits_, - audiochannels_); - } - return 0; - - CASE (SNDCTL_DSP_SETFMT) - { - int nBits; - switch (*intptr) - { - case AFMT_QUERY: - *intptr = audioformat_; - return 0; - break; - case AFMT_U16_BE: - case AFMT_U16_LE: - case AFMT_S16_BE: - case AFMT_S16_LE: - nBits = 16; - break; - case AFMT_U8: - case AFMT_S8: - nBits = 8; - break; - default: - nBits = 0; - } - if (nBits && IS_WRITE ()) - { - close_audio_out (); - if (audio_out_->query (audiofreq_, nBits, audiochannels_)) - { - audiobits_ = nBits; - audioformat_ = *intptr; - } - else - { - *intptr = audiobits_; - return -1; - } - } - if (nBits && IS_READ ()) - { - close_audio_in (); - if (audio_in_->query (audiofreq_, nBits, audiochannels_)) - { - audiobits_ = nBits; - audioformat_ = *intptr; - } - else - { - *intptr = audiobits_; - return -1; - } - } - return 0; - } - - CASE (SNDCTL_DSP_SPEED) - if (IS_WRITE ()) - { - close_audio_out (); - if (audio_out_->query (*intptr, audiobits_, audiochannels_)) - audiofreq_ = *intptr; - else - { - *intptr = audiofreq_; - return -1; - } - } - if (IS_READ ()) - { - close_audio_in (); - if (audio_in_->query (*intptr, audiobits_, audiochannels_)) - audiofreq_ = *intptr; - else - { - *intptr = audiofreq_; - return -1; - } - } - return 0; - - CASE (SNDCTL_DSP_STEREO) - { - int nChannels = *intptr + 1; - int res = ioctl (SNDCTL_DSP_CHANNELS, &nChannels); - *intptr = nChannels - 1; - return res; - } - - CASE (SNDCTL_DSP_CHANNELS) - { - int nChannels = *intptr; - - if (IS_WRITE ()) - { - close_audio_out (); - if (audio_out_->query (audiofreq_, audiobits_, nChannels)) - audiochannels_ = nChannels; - else - { - *intptr = audiochannels_; - return -1; - } - } - if (IS_READ ()) - { - close_audio_in (); - if (audio_in_->query (audiofreq_, audiobits_, nChannels)) - audiochannels_ = nChannels; - else - { - *intptr = audiochannels_; - return -1; - } - } - return 0; - } - - CASE (SNDCTL_DSP_GETOSPACE) - { - if (!IS_WRITE ()) - { - set_errno(EBADF); - return -1; - } - audio_buf_info *p = (audio_buf_info *) ptr; - audio_out_->buf_info (p, audiofreq_, audiobits_, audiochannels_); - debug_printf ("ptr=%p frags=%d fragsize=%d bytes=%d", - ptr, p->fragments, p->fragsize, p->bytes); - return 0; - } - - CASE (SNDCTL_DSP_GETISPACE) - { - if (!IS_READ ()) - { - set_errno(EBADF); - return -1; - } - audio_buf_info *p = (audio_buf_info *) ptr; - audio_in_->buf_info (p, audiofreq_, audiobits_, audiochannels_); - debug_printf ("ptr=%p frags=%d fragsize=%d bytes=%d", - ptr, p->fragments, p->fragsize, p->bytes); - return 0; - } - - CASE (SNDCTL_DSP_SETFRAGMENT) - // Fake!! esound & mikmod require this on non PowerPC platforms. - // - return 0; - - CASE (SNDCTL_DSP_GETFMTS) - *intptr = AFMT_S16_LE | AFMT_U8; // only native formats returned here - return 0; - - CASE (SNDCTL_DSP_GETCAPS) - *intptr = DSP_CAP_BATCH | DSP_CAP_DUPLEX; - return 0; - - CASE (SNDCTL_DSP_POST) - CASE (SNDCTL_DSP_SYNC) - // Stop audio out device - close_audio_out (); - // Stop audio in device - close_audio_in (); - return 0; - - default: - debug_printf ("/dev/dsp: ioctl 0x%08x not handled yet! FIXME:", cmd); - break; - -#undef CASE - }; - set_errno (EINVAL); - return -1; -} - -void -fhandler_dev_dsp::fixup_after_fork (HANDLE parent) -{ // called from new child process - debug_printf ("audio_in=%08x audio_out=%08x", - (int)audio_in_, (int)audio_out_); - if (archetype != this) - return ((fhandler_dev_dsp *)archetype)->fixup_after_fork (parent); - - if (audio_in_) - audio_in_ ->fork_fixup (parent); - if (audio_out_) - audio_out_->fork_fixup (parent); -} - -void -fhandler_dev_dsp::fixup_after_exec () -{ - debug_printf ("audio_in=%08x audio_out=%08x, close_on_exec %d", - (int) audio_in_, (int) audio_out_, close_on_exec ()); - if (!close_on_exec ()) - { - if (archetype != this) - return ((fhandler_dev_dsp *) archetype)->fixup_after_exec (); - - audio_in_ = NULL; - audio_out_ = NULL; - } -} |