Add efacf056cd

dockerdoom/trunk/pcsound/pcsound_sdl.c

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+// Emacs style mode select   -*- C++ -*- 
+//-----------------------------------------------------------------------------
+//
+// Copyright(C) 2007 Simon Howard
+//
+// This program is free software; you can redistribute it and/or
+// modify it under the terms of the GNU General Public License
+// as published by the Free Software Foundation; either version 2
+// of the License, or (at your option) any later version.
+//
+// This program 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 General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA
+// 02111-1307, USA.
+//
+// DESCRIPTION:
+//    PC speaker interface.
+//
+//-----------------------------------------------------------------------------
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "SDL.h"
+#include "SDL_mixer.h"
+
+#include "pcsound.h"
+#include "pcsound_internal.h"
+
+#define MAX_SOUND_SLICE_TIME 70 /* ms */
+#define SQUARE_WAVE_AMP 0x2000
+
+// If true, we initialized SDL and have the responsibility to shut it 
+// down
+
+static int sdl_was_initialized = 0;
+
+// Callback function to invoke when we want new sound data
+
+static pcsound_callback_func callback;
+
+// Output sound format
+
+static int mixing_freq;
+static Uint16 mixing_format;
+static int mixing_channels;
+
+// Currently playing sound
+// current_remaining is the number of remaining samples that must be played
+// before we invoke the callback to get the next frequency.
+
+static int current_remaining;
+static int current_freq;
+
+static int phase_offset = 0;
+
+// Mixer function that does the PC speaker emulation
+
+static void PCSound_Mix_Callback(void *udata, Uint8 *stream, int len)
+{
+    Sint16 *leftptr;
+    Sint16 *rightptr;
+    Sint16 this_value;
+    int oldfreq;
+    int i;
+    int nsamples;
+
+    // Number of samples is quadrupled, because of 16-bit and stereo
+
+    nsamples = len / 4;
+
+    leftptr = (Sint16 *) stream;
+    rightptr = ((Sint16 *) stream) + 1;
+    
+    // Fill the output buffer
+
+    for (i=0; i<nsamples; ++i)
+    {
+        // Has this sound expired? If so, invoke the callback to get 
+        // the next frequency.
+
+        while (current_remaining == 0) 
+        {
+            oldfreq = current_freq;
+
+            // Get the next frequency to play
+
+            callback(&current_remaining, &current_freq);
+
+            if (current_freq != 0)
+            {
+                // Adjust phase to match to the new frequency.
+                // This gives us a smooth transition between different tones,
+                // with no impulse changes.
+
+                phase_offset = (phase_offset * oldfreq) / current_freq;
+            }
+
+            current_remaining = (current_remaining * mixing_freq) / 1000;
+        }
+
+        // Set the value for this sample.
+        
+        if (current_freq == 0)
+        {
+            // Silence
+
+            this_value = 0;
+        }
+        else 
+        {
+            int frac;
+
+            // Determine whether we are at a peak or trough in the current
+            // sound.  Multiply by 2 so that frac % 2 will give 0 or 1 
+            // depending on whether we are at a peak or trough.
+
+            frac = (phase_offset * current_freq * 2) / mixing_freq;
+
+            if ((frac % 2) == 0) 
+            {
+                this_value = SQUARE_WAVE_AMP;
+            }
+            else
+            {
+                this_value = -SQUARE_WAVE_AMP;
+            }
+
+            ++phase_offset;
+        }
+
+        --current_remaining;
+
+        // Use the same value for the left and right channels.
+
+        *leftptr += this_value;
+        *rightptr += this_value;
+
+        leftptr += 2;
+        rightptr += 2;
+    }
+}
+
+static int SDLIsInitialized(void)
+{
+    int freq, channels;
+    Uint16 format;
+
+    return Mix_QuerySpec(&freq, &format, &channels);
+}
+
+static void PCSound_SDL_Shutdown(void)
+{
+    if (sdl_was_initialized)
+    {
+        Mix_CloseAudio();
+        SDL_QuitSubSystem(SDL_INIT_AUDIO);
+        sdl_was_initialized = 0;
+    }
+}
+
+// Calculate slice size, based on MAX_SOUND_SLICE_TIME.
+// The result must be a power of two.
+
+static int GetSliceSize(void)
+{
+    int limit;
+    int n;
+
+    limit = (pcsound_sample_rate * MAX_SOUND_SLICE_TIME) / 1000;
+
+    // Try all powers of two, not exceeding the limit.
+
+    for (n=0;; ++n)
+    {
+        // 2^n <= limit < 2^n+1 ?
+
+        if ((1 << (n + 1)) > limit)
+        {
+            return (1 << n);
+        }
+    }
+
+    // Should never happen?
+
+    return 1024;
+}
+
+static int PCSound_SDL_Init(pcsound_callback_func callback_func)
+{
+    int slicesize;
+
+    // Check if SDL_mixer has been opened already
+    // If not, we must initialize it now
+
+    if (!SDLIsInitialized())
+    {
+        if (SDL_Init(SDL_INIT_AUDIO) < 0)
+        {
+            fprintf(stderr, "Unable to set up sound.\n");
+            return 0;
+        }
+
+        slicesize = GetSliceSize();
+
+        if (Mix_OpenAudio(pcsound_sample_rate, AUDIO_S16SYS, 2, slicesize) < 0)
+        {
+            fprintf(stderr, "Error initializing SDL_mixer: %s\n", Mix_GetError());
+
+            SDL_QuitSubSystem(SDL_INIT_AUDIO);
+            return 0;
+        }
+
+        SDL_PauseAudio(0);
+
+        // When this module shuts down, it has the responsibility to 
+        // shut down SDL.
+
+        sdl_was_initialized = 1;
+    }
+
+    // Get the mixer frequency, format and number of channels.
+
+    Mix_QuerySpec(&mixing_freq, &mixing_format, &mixing_channels);
+
+    // Only supports AUDIO_S16SYS
+
+    if (mixing_format != AUDIO_S16SYS || mixing_channels != 2)
+    {
+        fprintf(stderr, 
+                "PCSound_SDL only supports native signed 16-bit LSB, "
+                "stereo format!\n");
+
+        PCSound_SDL_Shutdown();
+        return 0;
+    }
+
+    callback = callback_func;
+    current_freq = 0;
+    current_remaining = 0;
+
+    Mix_SetPostMix(PCSound_Mix_Callback, NULL);
+
+    return 1;
+}
+
+pcsound_driver_t pcsound_sdl_driver = 
+{
+    "SDL",
+    PCSound_SDL_Init,
+    PCSound_SDL_Shutdown,
+};
+