From a855d6202536ff28e5aae4f22a0f31d8f5b325d0 Mon Sep 17 00:00:00 2001 From: Franklin Wei Date: Sat, 21 Jan 2017 15:18:31 -0500 Subject: Port of Duke Nukem 3D This ports Fabien Sanglard's Chocolate Duke to run on a version of SDL for Rockbox. Change-Id: I8f2c4c78af19de10c1633ed7bb7a997b43256dd9 --- apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.c | 340 ++++++++++++++++++++++++++ 1 file changed, 340 insertions(+) create mode 100644 apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.c (limited to 'apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.c') diff --git a/apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.c b/apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.c new file mode 100644 index 0000000000..256c547f9b --- /dev/null +++ b/apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.c @@ -0,0 +1,340 @@ +/* + 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 + + Sam Lantinga + slouken@libsdl.org + + Modified in Oct 2004 by Hannu Savolainen + hannu@opensound.com +*/ +#include "SDL_config.h" + +/* Allow access to a raw mixing buffer */ + +#include /* For perror() */ +#include /* For strerror() */ +#include +#include +#include +#include +#include +#include +#include + +#if SDL_AUDIO_DRIVER_OSS_SOUNDCARD_H +/* This is installed on some systems */ +#include +#else +/* This is recommended by OSS */ +#include +#endif + +#include "SDL_timer.h" +#include "SDL_audio.h" +#include "../SDL_audiomem.h" +#include "../SDL_audio_c.h" +#include "../SDL_audiodev_c.h" +#include "SDL_dspaudio.h" + +/* The tag name used by DSP audio */ +#define DSP_DRIVER_NAME "dsp" + +/* Open the audio device for playback, and don't block if busy */ +#define OPEN_FLAGS (O_WRONLY|O_NONBLOCK) + +/* Audio driver functions */ +static int DSP_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void DSP_WaitAudio(_THIS); +static void DSP_PlayAudio(_THIS); +static Uint8 *DSP_GetAudioBuf(_THIS); +static void DSP_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 = DSP_OpenAudio; + this->WaitAudio = DSP_WaitAudio; + this->PlayAudio = DSP_PlayAudio; + this->GetAudioBuf = DSP_GetAudioBuf; + this->CloseAudio = DSP_CloseAudio; + + this->free = Audio_DeleteDevice; + + return this; +} + +AudioBootStrap DSP_bootstrap = { + DSP_DRIVER_NAME, "OSS /dev/dsp standard audio", + Audio_Available, Audio_CreateDevice +}; + +/* This function waits until it is possible to write a full sound buffer */ +static void DSP_WaitAudio(_THIS) +{ + /* Not needed at all since OSS handles waiting automagically */ +} + +static void DSP_PlayAudio(_THIS) +{ + if (write(audio_fd, mixbuf, mixlen)==-1) + { + perror("Audio write"); + this->enabled = 0; + } + +#ifdef DEBUG_AUDIO + fprintf(stderr, "Wrote %d bytes of audio data\n", mixlen); +#endif +} + +static Uint8 *DSP_GetAudioBuf(_THIS) +{ + return(mixbuf); +} + +static void DSP_CloseAudio(_THIS) +{ + if ( mixbuf != NULL ) { + SDL_FreeAudioMem(mixbuf); + mixbuf = NULL; + } + if ( audio_fd >= 0 ) { + close(audio_fd); + audio_fd = -1; + } +} + +static int DSP_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + char audiodev[1024]; + int format; + int value; + int frag_spec; + Uint16 test_format; + + /* Make sure fragment size stays a power of 2, or OSS fails. */ + /* I don't know which of these are actually legal values, though... */ + if (spec->channels > 8) + spec->channels = 8; + else if (spec->channels > 4) + spec->channels = 4; + else if (spec->channels > 2) + spec->channels = 2; + + /* 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); + } + mixbuf = NULL; + + /* Make the file descriptor use blocking writes with fcntl() */ + { long flags; + flags = fcntl(audio_fd, F_GETFL); + flags &= ~O_NONBLOCK; + if ( fcntl(audio_fd, F_SETFL, flags) < 0 ) { + SDL_SetError("Couldn't set audio blocking mode"); + DSP_CloseAudio(this); + return(-1); + } + } + + /* Get a list of supported hardware formats */ + if ( ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &value) < 0 ) { + perror("SNDCTL_DSP_GETFMTS"); + SDL_SetError("Couldn't get audio format list"); + DSP_CloseAudio(this); + return(-1); + } + + /* 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: + if ( value & AFMT_U8 ) { + format = AFMT_U8; + } + break; + case AUDIO_S16LSB: + if ( value & AFMT_S16_LE ) { + format = AFMT_S16_LE; + } + break; + case AUDIO_S16MSB: + if ( value & AFMT_S16_BE ) { + format = AFMT_S16_BE; + } + break; +#if 0 +/* + * These formats are not used by any real life systems so they are not + * needed here. + */ + case AUDIO_S8: + if ( value & AFMT_S8 ) { + format = AFMT_S8; + } + break; + case AUDIO_U16LSB: + if ( value & AFMT_U16_LE ) { + format = AFMT_U16_LE; + } + break; + case AUDIO_U16MSB: + if ( value & AFMT_U16_BE ) { + format = AFMT_U16_BE; + } + break; +#endif + default: + format = 0; + break; + } + if ( ! format ) { + test_format = SDL_NextAudioFormat(); + } + } + if ( format == 0 ) { + SDL_SetError("Couldn't find any hardware audio formats"); + DSP_CloseAudio(this); + return(-1); + } + spec->format = test_format; + + /* Set the audio format */ + value = format; + if ( (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) || + (value != format) ) { + perror("SNDCTL_DSP_SETFMT"); + SDL_SetError("Couldn't set audio format"); + DSP_CloseAudio(this); + return(-1); + } + + /* Set the number of channels of output */ + value = spec->channels; + if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &value) < 0 ) { + perror("SNDCTL_DSP_CHANNELS"); + SDL_SetError("Cannot set the number of channels"); + DSP_CloseAudio(this); + return(-1); + } + spec->channels = value; + + /* Set the DSP frequency */ + value = spec->freq; + if ( ioctl(audio_fd, SNDCTL_DSP_SPEED, &value) < 0 ) { + perror("SNDCTL_DSP_SPEED"); + SDL_SetError("Couldn't set audio frequency"); + DSP_CloseAudio(this); + return(-1); + } + spec->freq = value; + + /* Calculate the final parameters for this audio specification */ + SDL_CalculateAudioSpec(spec); + + /* Determine the power of two of the fragment size */ + for ( frag_spec = 0; (0x01U<size; ++frag_spec ); + if ( (0x01U<size ) { + SDL_SetError("Fragment size must be a power of two"); + DSP_CloseAudio(this); + return(-1); + } + frag_spec |= 0x00020000; /* two fragments, for low latency */ + + /* Set the audio buffering parameters */ +#ifdef DEBUG_AUDIO + fprintf(stderr, "Requesting %d fragments of size %d\n", + (frag_spec >> 16), 1<<(frag_spec&0xFFFF)); +#endif + if ( ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag_spec) < 0 ) { + perror("SNDCTL_DSP_SETFRAGMENT"); + } +#ifdef DEBUG_AUDIO + { audio_buf_info info; + ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &info); + fprintf(stderr, "fragments = %d\n", info.fragments); + fprintf(stderr, "fragstotal = %d\n", info.fragstotal); + fprintf(stderr, "fragsize = %d\n", info.fragsize); + fprintf(stderr, "bytes = %d\n", info.bytes); + } +#endif + + /* Allocate mixing buffer */ + mixlen = spec->size; + mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); + if ( mixbuf == NULL ) { + DSP_CloseAudio(this); + return(-1); + } + SDL_memset(mixbuf, spec->silence, spec->size); + + /* 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 v1.2.3