From d8239786b306ffda6d5d73753d01f0ad3358e1a0 Mon Sep 17 00:00:00 2001 From: Jesse Hall Date: Tue, 17 Jul 2012 16:58:55 -0700 Subject: Delete sdl-1.2.12 Change-Id: Ia96f80df04035ae84be3af468c945f2cec14f99c --- distrib/sdl-1.2.12/src/audio/paudio/SDL_paudio.c | 511 ----------------------- 1 file changed, 511 deletions(-) delete mode 100644 distrib/sdl-1.2.12/src/audio/paudio/SDL_paudio.c (limited to 'distrib/sdl-1.2.12/src/audio/paudio/SDL_paudio.c') diff --git a/distrib/sdl-1.2.12/src/audio/paudio/SDL_paudio.c b/distrib/sdl-1.2.12/src/audio/paudio/SDL_paudio.c deleted file mode 100644 index 7b07b59..0000000 --- a/distrib/sdl-1.2.12/src/audio/paudio/SDL_paudio.c +++ /dev/null @@ -1,511 +0,0 @@ -/* - SDL - Simple DirectMedia Layer - Copyright (C) 1997-2006 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 -#include -#include -#include -#include -#include - -#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 headers and probably others as well. - * I guess nobody ever uses audio... Shame over AIX header files. */ -#include -#undef BIG_ENDIAN -#include - -/* 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; -} - -- cgit v1.1