examples/alloopback.c

   1 /*
   2  * OpenAL Loopback Example
   3  *
   4  * Copyright (c) 2013 by Chris Robinson <chris.kcat@gmail.com>
   5  *
   6  * Permission is hereby granted, free of charge, to any person obtaining a copy
   7  * of this software and associated documentation files (the "Software"), to deal
   8  * in the Software without restriction, including without limitation the rights
   9  * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
  10  * copies of the Software, and to permit persons to whom the Software is
  11  * furnished to do so, subject to the following conditions:
  12  *
  13  * The above copyright notice and this permission notice shall be included in
  14  * all copies or substantial portions of the Software.
  15  *
  16  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
  17  * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
  18  * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
  19  * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
  20  * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
  21  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
  22  * THE SOFTWARE.
  23  */
  24
  25 /* This file contains an example for using the loopback device for custom
  26  * output handling.
  27  */
  28
  29 #include <assert.h>
  30 #include <math.h>
  31 #include <stdio.h>
  32
  33 #include "SDL.h"
  34 #include "SDL_audio.h"
  35 #include "SDL_error.h"
  36 #include "SDL_stdinc.h"
  37
  38 #include "AL/al.h"
  39 #include "AL/alc.h"
  40 #include "AL/alext.h"
  41
  42 #include "common/alhelpers.h"
  43
  44 #ifndef SDL_AUDIO_MASK_BITSIZE
  45 #define SDL_AUDIO_MASK_BITSIZE (0xFF)
  46 #endif
  47 #ifndef SDL_AUDIO_BITSIZE
  48 #define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE)
  49 #endif
  50
  51 #ifndef M_PI
  52 #define M_PI    (3.14159265358979323846)
  53 #endif
  54
  55 typedef struct {
  56     ALCdevice *Device;
  57     ALCcontext *Context;
  58
  59     ALCsizei FrameSize;
  60 } PlaybackInfo;
  61
  62 static LPALCLOOPBACKOPENDEVICESOFT alcLoopbackOpenDeviceSOFT;
  63 static LPALCISRENDERFORMATSUPPORTEDSOFT alcIsRenderFormatSupportedSOFT;
  64 static LPALCRENDERSAMPLESSOFT alcRenderSamplesSOFT;
  65
  66
  67 void SDLCALL RenderSDLSamples(void *userdata, Uint8 *stream, int len)
  68 {
  69     PlaybackInfo *playback = (PlaybackInfo*)userdata;
  70     alcRenderSamplesSOFT(playback->Device, stream, len/playback->FrameSize);
  71 }
  72
  73
  74 static const char *ChannelsName(ALCenum chans)
  75 {
  76     switch(chans)
  77     {
  78     case ALC_MONO_SOFT: return "Mono";
  79     case ALC_STEREO_SOFT: return "Stereo";
  80     case ALC_QUAD_SOFT: return "Quadraphonic";
  81     case ALC_5POINT1_SOFT: return "5.1 Surround";
  82     case ALC_6POINT1_SOFT: return "6.1 Surround";
  83     case ALC_7POINT1_SOFT: return "7.1 Surround";
  84     }
  85     return "Unknown Channels";
  86 }
  87
  88 static const char *TypeName(ALCenum type)
  89 {
  90     switch(type)
  91     {
  92     case ALC_BYTE_SOFT: return "S8";
  93     case ALC_UNSIGNED_BYTE_SOFT: return "U8";
  94     case ALC_SHORT_SOFT: return "S16";
  95     case ALC_UNSIGNED_SHORT_SOFT: return "U16";
  96     case ALC_INT_SOFT: return "S32";
  97     case ALC_UNSIGNED_INT_SOFT: return "U32";
  98     case ALC_FLOAT_SOFT: return "Float32";
  99     }
 100     return "Unknown Type";
 101 }
 102
 103 /* Creates a one second buffer containing a sine wave, and returns the new
 104  * buffer ID. */
 105 static ALuint CreateSineWave(void)
 106 {
 107     ALshort data[44100*4];
 108     ALuint buffer;
 109     ALenum err;
 110     ALuint i;
 111
 112     for(i = 0;i < 44100*4;i++)
 113         data[i] = (ALshort)(sin(i/44100.0 * 1000.0 * 2.0*M_PI) * 32767.0);
 114
 115     /* Buffer the audio data into a new buffer object. */
 116     buffer = 0;
 117     alGenBuffers(1, &buffer);
 118     alBufferData(buffer, AL_FORMAT_MONO16, data, sizeof(data), 44100);
 119
 120     /* Check if an error occured, and clean up if so. */
 121     err = alGetError();
 122     if(err != AL_NO_ERROR)
 123     {
 124         fprintf(stderr, "OpenAL Error: %s\n", alGetString(err));
 125         if(alIsBuffer(buffer))
 126             alDeleteBuffers(1, &buffer);
 127         return 0;
 128     }
 129
 130     return buffer;
 131 }
 132
 133
 134 int main(int argc, char *argv[])
 135 {
 136     PlaybackInfo playback = { NULL, NULL, 0 };
 137     SDL_AudioSpec desired, obtained;
 138     ALuint source, buffer;
 139     ALCint attrs[16];
 140     ALenum state;
 141     (void)argc;
 142     (void)argv;
 143
 144     /* Print out error if extension is missing. */
 145     if(!alcIsExtensionPresent(NULL, "ALC_SOFT_loopback"))
 146     {
 147         fprintf(stderr, "Error: ALC_SOFT_loopback not supported!\n");
 148         return 1;
 149     }
 150
 151     /* Define a macro to help load the function pointers. */
 152 #define LOAD_PROC(T, x)  ((x) = (T)alcGetProcAddress(NULL, #x))
 153     LOAD_PROC(LPALCLOOPBACKOPENDEVICESOFT, alcLoopbackOpenDeviceSOFT);
 154     LOAD_PROC(LPALCISRENDERFORMATSUPPORTEDSOFT, alcIsRenderFormatSupportedSOFT);
 155     LOAD_PROC(LPALCRENDERSAMPLESSOFT, alcRenderSamplesSOFT);
 156 #undef LOAD_PROC
 157
 158     if(SDL_Init(SDL_INIT_AUDIO) == -1)
 159     {
 160         fprintf(stderr, "Failed to init SDL audio: %s\n", SDL_GetError());
 161         return 1;
 162     }
 163
 164     /* Set up SDL audio with our requested format and callback. */
 165     desired.channels = 2;
 166     desired.format = AUDIO_S16SYS;
 167     desired.freq = 44100;
 168     desired.padding = 0;
 169     desired.samples = 4096;
 170     desired.callback = RenderSDLSamples;
 171     desired.userdata = &playback;
 172     if(SDL_OpenAudio(&desired, &obtained) != 0)
 173     {
 174         SDL_Quit();
 175         fprintf(stderr, "Failed to open SDL audio: %s\n", SDL_GetError());
 176         return 1;
 177     }
 178
 179     /* Set up our OpenAL attributes based on what we got from SDL. */
 180     attrs[0] = ALC_FORMAT_CHANNELS_SOFT;
 181     if(obtained.channels == 1)
 182         attrs[1] = ALC_MONO_SOFT;
 183     else if(obtained.channels == 2)
 184         attrs[1] = ALC_STEREO_SOFT;
 185     else
 186     {
 187         fprintf(stderr, "Unhandled SDL channel count: %d\n", obtained.channels);
 188         goto error;
 189     }
 190
 191     attrs[2] = ALC_FORMAT_TYPE_SOFT;
 192     if(obtained.format == AUDIO_U8)
 193         attrs[3] = ALC_UNSIGNED_BYTE_SOFT;
 194     else if(obtained.format == AUDIO_S8)
 195         attrs[3] = ALC_BYTE_SOFT;
 196     else if(obtained.format == AUDIO_U16SYS)
 197         attrs[3] = ALC_UNSIGNED_SHORT_SOFT;
 198     else if(obtained.format == AUDIO_S16SYS)
 199         attrs[3] = ALC_SHORT_SOFT;
 200     else
 201     {
 202         fprintf(stderr, "Unhandled SDL format: 0x%04x\n", obtained.format);
 203         goto error;
 204     }
 205
 206     attrs[4] = ALC_FREQUENCY;
 207     attrs[5] = obtained.freq;
 208
 209     attrs[6] = 0; /* end of list */
 210
 211     playback.FrameSize = obtained.channels * SDL_AUDIO_BITSIZE(obtained.format) / 8;
 212
 213     /* Initialize OpenAL loopback device, using our format attributes. */
 214     playback.Device = alcLoopbackOpenDeviceSOFT(NULL);
 215     if(!playback.Device)
 216     {
 217         fprintf(stderr, "Failed to open loopback device!\n");
 218         goto error;
 219     }
 220     /* Make sure the format is supported before setting them on the device. */
 221     if(alcIsRenderFormatSupportedSOFT(playback.Device, attrs[5], attrs[1], attrs[3]) == ALC_FALSE)
 222     {
 223         fprintf(stderr, "Render format not supported: %s, %s, %dhz\n",
 224                         ChannelsName(attrs[1]), TypeName(attrs[3]), attrs[5]);
 225         goto error;
 226     }
 227     playback.Context = alcCreateContext(playback.Device, attrs);
 228     if(!playback.Context || alcMakeContextCurrent(playback.Context) == ALC_FALSE)
 229     {
 230         fprintf(stderr, "Failed to set an OpenAL audio context\n");
 231         goto error;
 232     }
 233
 234     /* Start SDL playing. Our callback (thus alcRenderSamplesSOFT) will now
 235      * start being called regularly to update the AL playback state. */
 236     SDL_PauseAudio(0);
 237
 238     /* Load the sound into a buffer. */
 239     buffer = CreateSineWave();
 240     if(!buffer)
 241     {
 242         SDL_CloseAudio();
 243         alcDestroyContext(playback.Context);
 244         alcCloseDevice(playback.Device);
 245         SDL_Quit();
 246         return 1;
 247     }
 248
 249     /* Create the source to play the sound with. */
 250     source = 0;
 251     alGenSources(1, &source);
 252     alSourcei(source, AL_BUFFER, (ALint)buffer);
 253     assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
 254
 255     /* Play the sound until it finishes. */
 256     alSourcePlay(source);
 257     do {
 258         al_nssleep(10000000);
 259         alGetSourcei(source, AL_SOURCE_STATE, &state);
 260     } while(alGetError() == AL_NO_ERROR && state == AL_PLAYING);
 261
 262     /* All done. Delete resources, and close OpenAL. */
 263     alDeleteSources(1, &source);
 264     alDeleteBuffers(1, &buffer);
 265
 266     /* Stop SDL playing. */
 267     SDL_PauseAudio(1);
 268
 269     /* Close up OpenAL and SDL. */
 270     SDL_CloseAudio();
 271     alcDestroyContext(playback.Context);
 272     alcCloseDevice(playback.Device);
 273     SDL_Quit();
 274
 275     return 0;
 276
 277 error:
 278     SDL_CloseAudio();
 279     if(playback.Context)
 280         alcDestroyContext(playback.Context);
 281     if(playback.Device)
 282         alcCloseDevice(playback.Device);
 283     SDL_Quit();
 284
 285     return 1;
 286 }