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Diffstat (limited to 'distrib/sdl-1.2.15/src/audio/paudio/SDL_paudio.c')
-rw-r--r--distrib/sdl-1.2.15/src/audio/paudio/SDL_paudio.c511
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diff --git a/distrib/sdl-1.2.15/src/audio/paudio/SDL_paudio.c b/distrib/sdl-1.2.15/src/audio/paudio/SDL_paudio.c
new file mode 100644
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+++ b/distrib/sdl-1.2.15/src/audio/paudio/SDL_paudio.c
@@ -0,0 +1,511 @@
+/*
+ SDL - Simple DirectMedia Layer
+ Copyright (C) 1997-2012 Sam Lantinga
+
+ This library is free software; you can redistribute it and/or
+ modify it under the terms of the GNU Lesser General Public
+ License as published by the Free Software Foundation; either
+ version 2.1 of the License, or (at your option) any later version.
+
+ This library is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ Lesser General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public
+ License along with this library; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+
+ Carsten Griwodz
+ griff@kom.tu-darmstadt.de
+
+ based on linux/SDL_dspaudio.c by Sam Lantinga
+*/
+#include "SDL_config.h"
+
+/* Allow access to a raw mixing buffer */
+
+#include <errno.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/time.h>
+#include <sys/ioctl.h>
+#include <sys/stat.h>
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "../SDL_audiomem.h"
+#include "../SDL_audio_c.h"
+#include "../SDL_audiodev_c.h"
+#include "SDL_paudio.h"
+
+#define DEBUG_AUDIO 1
+
+/* A conflict within AIX 4.3.3 <sys/> headers and probably others as well.
+ * I guess nobody ever uses audio... Shame over AIX header files. */
+#include <sys/machine.h>
+#undef BIG_ENDIAN
+#include <sys/audio.h>
+
+/* The tag name used by paud audio */
+#define Paud_DRIVER_NAME "paud"
+
+/* Open the audio device for playback, and don't block if busy */
+/* #define OPEN_FLAGS (O_WRONLY|O_NONBLOCK) */
+#define OPEN_FLAGS O_WRONLY
+
+/* Audio driver functions */
+static int Paud_OpenAudio(_THIS, SDL_AudioSpec *spec);
+static void Paud_WaitAudio(_THIS);
+static void Paud_PlayAudio(_THIS);
+static Uint8 *Paud_GetAudioBuf(_THIS);
+static void Paud_CloseAudio(_THIS);
+
+/* Audio driver bootstrap functions */
+
+static int Audio_Available(void)
+{
+ int fd;
+ int available;
+
+ available = 0;
+ fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 0);
+ if ( fd >= 0 ) {
+ available = 1;
+ close(fd);
+ }
+ return(available);
+}
+
+static void Audio_DeleteDevice(SDL_AudioDevice *device)
+{
+ SDL_free(device->hidden);
+ SDL_free(device);
+}
+
+static SDL_AudioDevice *Audio_CreateDevice(int devindex)
+{
+ SDL_AudioDevice *this;
+
+ /* Initialize all variables that we clean on shutdown */
+ this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
+ if ( this ) {
+ SDL_memset(this, 0, (sizeof *this));
+ this->hidden = (struct SDL_PrivateAudioData *)
+ SDL_malloc((sizeof *this->hidden));
+ }
+ if ( (this == NULL) || (this->hidden == NULL) ) {
+ SDL_OutOfMemory();
+ if ( this ) {
+ SDL_free(this);
+ }
+ return(0);
+ }
+ SDL_memset(this->hidden, 0, (sizeof *this->hidden));
+ audio_fd = -1;
+
+ /* Set the function pointers */
+ this->OpenAudio = Paud_OpenAudio;
+ this->WaitAudio = Paud_WaitAudio;
+ this->PlayAudio = Paud_PlayAudio;
+ this->GetAudioBuf = Paud_GetAudioBuf;
+ this->CloseAudio = Paud_CloseAudio;
+
+ this->free = Audio_DeleteDevice;
+
+ return this;
+}
+
+AudioBootStrap Paud_bootstrap = {
+ Paud_DRIVER_NAME, "AIX Paudio",
+ Audio_Available, Audio_CreateDevice
+};
+
+/* This function waits until it is possible to write a full sound buffer */
+static void Paud_WaitAudio(_THIS)
+{
+ fd_set fdset;
+
+ /* See if we need to use timed audio synchronization */
+ if ( frame_ticks ) {
+ /* Use timer for general audio synchronization */
+ Sint32 ticks;
+
+ ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS;
+ if ( ticks > 0 ) {
+ SDL_Delay(ticks);
+ }
+ } else {
+ audio_buffer paud_bufinfo;
+
+ /* Use select() for audio synchronization */
+ struct timeval timeout;
+ FD_ZERO(&fdset);
+ FD_SET(audio_fd, &fdset);
+
+ if ( ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0 ) {
+#ifdef DEBUG_AUDIO
+ fprintf(stderr, "Couldn't get audio buffer information\n");
+#endif
+ timeout.tv_sec = 10;
+ timeout.tv_usec = 0;
+ } else {
+ long ms_in_buf = paud_bufinfo.write_buf_time;
+ timeout.tv_sec = ms_in_buf/1000;
+ ms_in_buf = ms_in_buf - timeout.tv_sec*1000;
+ timeout.tv_usec = ms_in_buf*1000;
+#ifdef DEBUG_AUDIO
+ fprintf( stderr,
+ "Waiting for write_buf_time=%ld,%ld\n",
+ timeout.tv_sec,
+ timeout.tv_usec );
+#endif
+ }
+
+#ifdef DEBUG_AUDIO
+ fprintf(stderr, "Waiting for audio to get ready\n");
+#endif
+ if ( select(audio_fd+1, NULL, &fdset, NULL, &timeout) <= 0 ) {
+ const char *message = "Audio timeout - buggy audio driver? (disabled)";
+ /*
+ * In general we should never print to the screen,
+ * but in this case we have no other way of letting
+ * the user know what happened.
+ */
+ fprintf(stderr, "SDL: %s - %s\n", strerror(errno), message);
+ this->enabled = 0;
+ /* Don't try to close - may hang */
+ audio_fd = -1;
+#ifdef DEBUG_AUDIO
+ fprintf(stderr, "Done disabling audio\n");
+#endif
+ }
+#ifdef DEBUG_AUDIO
+ fprintf(stderr, "Ready!\n");
+#endif
+ }
+}
+
+static void Paud_PlayAudio(_THIS)
+{
+ int written;
+
+ /* Write the audio data, checking for EAGAIN on broken audio drivers */
+ do {
+ written = write(audio_fd, mixbuf, mixlen);
+ if ( (written < 0) && ((errno == 0) || (errno == EAGAIN)) ) {
+ SDL_Delay(1); /* Let a little CPU time go by */
+ }
+ } while ( (written < 0) &&
+ ((errno == 0) || (errno == EAGAIN) || (errno == EINTR)) );
+
+ /* If timer synchronization is enabled, set the next write frame */
+ if ( frame_ticks ) {
+ next_frame += frame_ticks;
+ }
+
+ /* If we couldn't write, assume fatal error for now */
+ if ( written < 0 ) {
+ this->enabled = 0;
+ }
+#ifdef DEBUG_AUDIO
+ fprintf(stderr, "Wrote %d bytes of audio data\n", written);
+#endif
+}
+
+static Uint8 *Paud_GetAudioBuf(_THIS)
+{
+ return mixbuf;
+}
+
+static void Paud_CloseAudio(_THIS)
+{
+ if ( mixbuf != NULL ) {
+ SDL_FreeAudioMem(mixbuf);
+ mixbuf = NULL;
+ }
+ if ( audio_fd >= 0 ) {
+ close(audio_fd);
+ audio_fd = -1;
+ }
+}
+
+static int Paud_OpenAudio(_THIS, SDL_AudioSpec *spec)
+{
+ char audiodev[1024];
+ int format;
+ int bytes_per_sample;
+ Uint16 test_format;
+ audio_init paud_init;
+ audio_buffer paud_bufinfo;
+ audio_status paud_status;
+ audio_control paud_control;
+ audio_change paud_change;
+
+ /* Reset the timer synchronization flag */
+ frame_ticks = 0.0;
+
+ /* Open the audio device */
+ audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0);
+ if ( audio_fd < 0 ) {
+ SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
+ return -1;
+ }
+
+ /*
+ * We can't set the buffer size - just ask the device for the maximum
+ * that we can have.
+ */
+ if ( ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0 ) {
+ SDL_SetError("Couldn't get audio buffer information");
+ return -1;
+ }
+
+ mixbuf = NULL;
+
+ if ( spec->channels > 1 )
+ spec->channels = 2;
+ else
+ spec->channels = 1;
+
+ /*
+ * Fields in the audio_init structure:
+ *
+ * Ignored by us:
+ *
+ * paud.loadpath[LOAD_PATH]; * DSP code to load, MWave chip only?
+ * paud.slot_number; * slot number of the adapter
+ * paud.device_id; * adapter identification number
+ *
+ * Input:
+ *
+ * paud.srate; * the sampling rate in Hz
+ * paud.bits_per_sample; * 8, 16, 32, ...
+ * paud.bsize; * block size for this rate
+ * paud.mode; * ADPCM, PCM, MU_LAW, A_LAW, SOURCE_MIX
+ * paud.channels; * 1=mono, 2=stereo
+ * paud.flags; * FIXED - fixed length data
+ * * LEFT_ALIGNED, RIGHT_ALIGNED (var len only)
+ * * TWOS_COMPLEMENT - 2's complement data
+ * * SIGNED - signed? comment seems wrong in sys/audio.h
+ * * BIG_ENDIAN
+ * paud.operation; * PLAY, RECORD
+ *
+ * Output:
+ *
+ * paud.flags; * PITCH - pitch is supported
+ * * INPUT - input is supported
+ * * OUTPUT - output is supported
+ * * MONITOR - monitor is supported
+ * * VOLUME - volume is supported
+ * * VOLUME_DELAY - volume delay is supported
+ * * BALANCE - balance is supported
+ * * BALANCE_DELAY - balance delay is supported
+ * * TREBLE - treble control is supported
+ * * BASS - bass control is supported
+ * * BESTFIT_PROVIDED - best fit returned
+ * * LOAD_CODE - DSP load needed
+ * paud.rc; * NO_PLAY - DSP code can't do play requests
+ * * NO_RECORD - DSP code can't do record requests
+ * * INVALID_REQUEST - request was invalid
+ * * CONFLICT - conflict with open's flags
+ * * OVERLOADED - out of DSP MIPS or memory
+ * paud.position_resolution; * smallest increment for position
+ */
+
+ paud_init.srate = spec->freq;
+ paud_init.mode = PCM;
+ paud_init.operation = PLAY;
+ paud_init.channels = spec->channels;
+
+ /* Try for a closest match on audio format */
+ format = 0;
+ for ( test_format = SDL_FirstAudioFormat(spec->format);
+ ! format && test_format; ) {
+#ifdef DEBUG_AUDIO
+ fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
+#endif
+ switch ( test_format ) {
+ case AUDIO_U8:
+ bytes_per_sample = 1;
+ paud_init.bits_per_sample = 8;
+ paud_init.flags = TWOS_COMPLEMENT | FIXED;
+ format = 1;
+ break;
+ case AUDIO_S8:
+ bytes_per_sample = 1;
+ paud_init.bits_per_sample = 8;
+ paud_init.flags = SIGNED |
+ TWOS_COMPLEMENT | FIXED;
+ format = 1;
+ break;
+ case AUDIO_S16LSB:
+ bytes_per_sample = 2;
+ paud_init.bits_per_sample = 16;
+ paud_init.flags = SIGNED |
+ TWOS_COMPLEMENT | FIXED;
+ format = 1;
+ break;
+ case AUDIO_S16MSB:
+ bytes_per_sample = 2;
+ paud_init.bits_per_sample = 16;
+ paud_init.flags = BIG_ENDIAN |
+ SIGNED |
+ TWOS_COMPLEMENT | FIXED;
+ format = 1;
+ break;
+ case AUDIO_U16LSB:
+ bytes_per_sample = 2;
+ paud_init.bits_per_sample = 16;
+ paud_init.flags = TWOS_COMPLEMENT | FIXED;
+ format = 1;
+ break;
+ case AUDIO_U16MSB:
+ bytes_per_sample = 2;
+ paud_init.bits_per_sample = 16;
+ paud_init.flags = BIG_ENDIAN |
+ TWOS_COMPLEMENT | FIXED;
+ format = 1;
+ break;
+ default:
+ break;
+ }
+ if ( ! format ) {
+ test_format = SDL_NextAudioFormat();
+ }
+ }
+ if ( format == 0 ) {
+#ifdef DEBUG_AUDIO
+ fprintf(stderr, "Couldn't find any hardware audio formats\n");
+#endif
+ SDL_SetError("Couldn't find any hardware audio formats");
+ return -1;
+ }
+ spec->format = test_format;
+
+ /*
+ * We know the buffer size and the max number of subsequent writes
+ * that can be pending. If more than one can pend, allow the application
+ * to do something like double buffering between our write buffer and
+ * the device's own buffer that we are filling with write() anyway.
+ *
+ * We calculate spec->samples like this because SDL_CalculateAudioSpec()
+ * will give put paud_bufinfo.write_buf_cap (or paud_bufinfo.write_buf_cap/2)
+ * into spec->size in return.
+ */
+ if ( paud_bufinfo.request_buf_cap == 1 )
+ {
+ spec->samples = paud_bufinfo.write_buf_cap
+ / bytes_per_sample
+ / spec->channels;
+ }
+ else
+ {
+ spec->samples = paud_bufinfo.write_buf_cap
+ / bytes_per_sample
+ / spec->channels
+ / 2;
+ }
+ paud_init.bsize = bytes_per_sample * spec->channels;
+
+ SDL_CalculateAudioSpec(spec);
+
+ /*
+ * The AIX paud device init can't modify the values of the audio_init
+ * structure that we pass to it. So we don't need any recalculation
+ * of this stuff and no reinit call as in linux dsp and dma code.
+ *
+ * /dev/paud supports all of the encoding formats, so we don't need
+ * to do anything like reopening the device, either.
+ */
+ if ( ioctl(audio_fd, AUDIO_INIT, &paud_init) < 0 ) {
+ switch ( paud_init.rc )
+ {
+ case 1 :
+ SDL_SetError("Couldn't set audio format: DSP can't do play requests");
+ return -1;
+ break;
+ case 2 :
+ SDL_SetError("Couldn't set audio format: DSP can't do record requests");
+ return -1;
+ break;
+ case 4 :
+ SDL_SetError("Couldn't set audio format: request was invalid");
+ return -1;
+ break;
+ case 5 :
+ SDL_SetError("Couldn't set audio format: conflict with open's flags");
+ return -1;
+ break;
+ case 6 :
+ SDL_SetError("Couldn't set audio format: out of DSP MIPS or memory");
+ return -1;
+ break;
+ default :
+ SDL_SetError("Couldn't set audio format: not documented in sys/audio.h");
+ return -1;
+ break;
+ }
+ }
+
+ /* Allocate mixing buffer */
+ mixlen = spec->size;
+ mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
+ if ( mixbuf == NULL ) {
+ return -1;
+ }
+ SDL_memset(mixbuf, spec->silence, spec->size);
+
+ /*
+ * Set some paramters: full volume, first speaker that we can find.
+ * Ignore the other settings for now.
+ */
+ paud_change.input = AUDIO_IGNORE; /* the new input source */
+ paud_change.output = OUTPUT_1; /* EXTERNAL_SPEAKER,INTERNAL_SPEAKER,OUTPUT_1 */
+ paud_change.monitor = AUDIO_IGNORE; /* the new monitor state */
+ paud_change.volume = 0x7fffffff; /* volume level [0-0x7fffffff] */
+ paud_change.volume_delay = AUDIO_IGNORE; /* the new volume delay */
+ paud_change.balance = 0x3fffffff; /* the new balance */
+ paud_change.balance_delay = AUDIO_IGNORE; /* the new balance delay */
+ paud_change.treble = AUDIO_IGNORE; /* the new treble state */
+ paud_change.bass = AUDIO_IGNORE; /* the new bass state */
+ paud_change.pitch = AUDIO_IGNORE; /* the new pitch state */
+
+ paud_control.ioctl_request = AUDIO_CHANGE;
+ paud_control.request_info = (char*)&paud_change;
+ if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 ) {
+#ifdef DEBUG_AUDIO
+ fprintf(stderr, "Can't change audio display settings\n" );
+#endif
+ }
+
+ /*
+ * Tell the device to expect data. Actual start will wait for
+ * the first write() call.
+ */
+ paud_control.ioctl_request = AUDIO_START;
+ paud_control.position = 0;
+ if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 ) {
+#ifdef DEBUG_AUDIO
+ fprintf(stderr, "Can't start audio play\n" );
+#endif
+ SDL_SetError("Can't start audio play");
+ return -1;
+ }
+
+ /* Check to see if we need to use select() workaround */
+ { char *workaround;
+ workaround = SDL_getenv("SDL_DSP_NOSELECT");
+ if ( workaround ) {
+ frame_ticks = (float)(spec->samples*1000)/spec->freq;
+ next_frame = SDL_GetTicks()+frame_ticks;
+ }
+ }
+
+ /* Get the parent process id (we're the parent of the audio thread) */
+ parent = getpid();
+
+ /* We're ready to rock and roll. :-) */
+ return 0;
+}
+