| /* |
| SDL - Simple DirectMedia Layer |
| Copyright (C) 1997-2009 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; |
| } |
| |
