This change rewrites and simplifies the copyright headers at the top of all source files: * Remove "Emacs style mode select" line; this line was included in the headers for the originally released source files and appears to be to set the file type for old versions of Emacs. I'm not sure entirely why it was required but I don't think it is any more. * Remove "You should have received a copy of..." text from copyright header. This refers to the old 59 Temple Place address where the FSF headquarters used to be located and is no longer correct. Rather than change to the new address, just remove the paragraph as it is superfluous anyway. This fixes #311. * Remove ---- separator lines so that the file headers are barer and more simplified.
250 lines
5.6 KiB
C
250 lines
5.6 KiB
C
//
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// Copyright(C) 2005-2014 Simon Howard
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//
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// This program is free software; you can redistribute it and/or
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// modify it under the terms of the GNU General Public License
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// as published by the Free Software Foundation; either version 2
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// of the License, or (at your option) any later version.
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//
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// This program is distributed in the hope that it will be useful,
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// but WITHOUT ANY WARRANTY; without even the implied warranty of
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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// GNU General Public License for more details.
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//
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// DESCRIPTION:
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// PC speaker interface.
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//
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#include <stdio.h>
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#include <stdlib.h>
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#include "SDL.h"
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#include "SDL_mixer.h"
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#include "pcsound.h"
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#include "pcsound_internal.h"
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#define MAX_SOUND_SLICE_TIME 70 /* ms */
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#define SQUARE_WAVE_AMP 0x2000
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// If true, we initialized SDL and have the responsibility to shut it
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// down
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static int sdl_was_initialized = 0;
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// Callback function to invoke when we want new sound data
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static pcsound_callback_func callback;
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// Output sound format
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static int mixing_freq;
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static Uint16 mixing_format;
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static int mixing_channels;
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// Currently playing sound
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// current_remaining is the number of remaining samples that must be played
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// before we invoke the callback to get the next frequency.
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static int current_remaining;
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static int current_freq;
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static int phase_offset = 0;
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// Mixer function that does the PC speaker emulation
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static void PCSound_Mix_Callback(void *udata, Uint8 *stream, int len)
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{
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Sint16 *leftptr;
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Sint16 *rightptr;
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Sint16 this_value;
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int oldfreq;
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int i;
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int nsamples;
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// Number of samples is quadrupled, because of 16-bit and stereo
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nsamples = len / 4;
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leftptr = (Sint16 *) stream;
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rightptr = ((Sint16 *) stream) + 1;
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// Fill the output buffer
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for (i=0; i<nsamples; ++i)
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{
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// Has this sound expired? If so, invoke the callback to get
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// the next frequency.
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while (current_remaining == 0)
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{
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oldfreq = current_freq;
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// Get the next frequency to play
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callback(¤t_remaining, ¤t_freq);
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if (current_freq != 0)
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{
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// Adjust phase to match to the new frequency.
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// This gives us a smooth transition between different tones,
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// with no impulse changes.
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phase_offset = (phase_offset * oldfreq) / current_freq;
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}
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current_remaining = (current_remaining * mixing_freq) / 1000;
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}
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// Set the value for this sample.
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if (current_freq == 0)
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{
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// Silence
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this_value = 0;
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}
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else
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{
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int frac;
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// Determine whether we are at a peak or trough in the current
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// sound. Multiply by 2 so that frac % 2 will give 0 or 1
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// depending on whether we are at a peak or trough.
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frac = (phase_offset * current_freq * 2) / mixing_freq;
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if ((frac % 2) == 0)
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{
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this_value = SQUARE_WAVE_AMP;
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}
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else
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{
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this_value = -SQUARE_WAVE_AMP;
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}
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++phase_offset;
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}
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--current_remaining;
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// Use the same value for the left and right channels.
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*leftptr += this_value;
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*rightptr += this_value;
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leftptr += 2;
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rightptr += 2;
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}
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}
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static int SDLIsInitialized(void)
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{
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int freq, channels;
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Uint16 format;
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return Mix_QuerySpec(&freq, &format, &channels);
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}
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static void PCSound_SDL_Shutdown(void)
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{
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if (sdl_was_initialized)
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{
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Mix_CloseAudio();
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SDL_QuitSubSystem(SDL_INIT_AUDIO);
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sdl_was_initialized = 0;
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}
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}
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// Calculate slice size, based on MAX_SOUND_SLICE_TIME.
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// The result must be a power of two.
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static int GetSliceSize(void)
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{
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int limit;
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int n;
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limit = (pcsound_sample_rate * MAX_SOUND_SLICE_TIME) / 1000;
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// Try all powers of two, not exceeding the limit.
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for (n=0;; ++n)
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{
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// 2^n <= limit < 2^n+1 ?
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if ((1 << (n + 1)) > limit)
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{
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return (1 << n);
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}
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}
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// Should never happen?
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return 1024;
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}
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static int PCSound_SDL_Init(pcsound_callback_func callback_func)
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{
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int slicesize;
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// Check if SDL_mixer has been opened already
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// If not, we must initialize it now
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if (!SDLIsInitialized())
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{
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if (SDL_Init(SDL_INIT_AUDIO) < 0)
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{
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fprintf(stderr, "Unable to set up sound.\n");
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return 0;
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}
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slicesize = GetSliceSize();
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if (Mix_OpenAudio(pcsound_sample_rate, AUDIO_S16SYS, 2, slicesize) < 0)
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{
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fprintf(stderr, "Error initializing SDL_mixer: %s\n", Mix_GetError());
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SDL_QuitSubSystem(SDL_INIT_AUDIO);
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return 0;
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}
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SDL_PauseAudio(0);
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// When this module shuts down, it has the responsibility to
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// shut down SDL.
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sdl_was_initialized = 1;
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}
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// Get the mixer frequency, format and number of channels.
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Mix_QuerySpec(&mixing_freq, &mixing_format, &mixing_channels);
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// Only supports AUDIO_S16SYS
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if (mixing_format != AUDIO_S16SYS || mixing_channels != 2)
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{
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fprintf(stderr,
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"PCSound_SDL only supports native signed 16-bit LSB, "
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"stereo format!\n");
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PCSound_SDL_Shutdown();
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return 0;
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}
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callback = callback_func;
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current_freq = 0;
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current_remaining = 0;
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Mix_SetPostMix(PCSound_Mix_Callback, NULL);
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return 1;
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}
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pcsound_driver_t pcsound_sdl_driver =
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{
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"SDL",
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PCSound_SDL_Init,
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PCSound_SDL_Shutdown,
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};
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