🛠️🐜 Antkeeper superbuild with dependencies included https://antkeeper.com
You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
 
 
 
 
 
 

283 lines
8.1 KiB

/*
* OpenAL Loopback Example
*
* Copyright (c) 2013 by Chris Robinson <chris.kcat@gmail.com>
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/* This file contains an example for using the loopback device for custom
* output handling.
*/
#include <stdio.h>
#include <assert.h>
#include <math.h>
#include <SDL.h>
#include "AL/al.h"
#include "AL/alc.h"
#include "AL/alext.h"
#include "common/alhelpers.h"
#ifndef SDL_AUDIO_MASK_BITSIZE
#define SDL_AUDIO_MASK_BITSIZE (0xFF)
#endif
#ifndef SDL_AUDIO_BITSIZE
#define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE)
#endif
#ifndef M_PI
#define M_PI (3.14159265358979323846)
#endif
typedef struct {
ALCdevice *Device;
ALCcontext *Context;
ALCsizei FrameSize;
} PlaybackInfo;
static LPALCLOOPBACKOPENDEVICESOFT alcLoopbackOpenDeviceSOFT;
static LPALCISRENDERFORMATSUPPORTEDSOFT alcIsRenderFormatSupportedSOFT;
static LPALCRENDERSAMPLESSOFT alcRenderSamplesSOFT;
void SDLCALL RenderSDLSamples(void *userdata, Uint8 *stream, int len)
{
PlaybackInfo *playback = (PlaybackInfo*)userdata;
alcRenderSamplesSOFT(playback->Device, stream, len/playback->FrameSize);
}
static const char *ChannelsName(ALCenum chans)
{
switch(chans)
{
case ALC_MONO_SOFT: return "Mono";
case ALC_STEREO_SOFT: return "Stereo";
case ALC_QUAD_SOFT: return "Quadraphonic";
case ALC_5POINT1_SOFT: return "5.1 Surround";
case ALC_6POINT1_SOFT: return "6.1 Surround";
case ALC_7POINT1_SOFT: return "7.1 Surround";
}
return "Unknown Channels";
}
static const char *TypeName(ALCenum type)
{
switch(type)
{
case ALC_BYTE_SOFT: return "S8";
case ALC_UNSIGNED_BYTE_SOFT: return "U8";
case ALC_SHORT_SOFT: return "S16";
case ALC_UNSIGNED_SHORT_SOFT: return "U16";
case ALC_INT_SOFT: return "S32";
case ALC_UNSIGNED_INT_SOFT: return "U32";
case ALC_FLOAT_SOFT: return "Float32";
}
return "Unknown Type";
}
/* Creates a one second buffer containing a sine wave, and returns the new
* buffer ID. */
static ALuint CreateSineWave(void)
{
ALshort data[44100*4];
ALuint buffer;
ALenum err;
ALuint i;
for(i = 0;i < 44100*4;i++)
data[i] = (ALshort)(sin(i/44100.0 * 1000.0 * 2.0*M_PI) * 32767.0);
/* Buffer the audio data into a new buffer object. */
buffer = 0;
alGenBuffers(1, &buffer);
alBufferData(buffer, AL_FORMAT_MONO16, data, sizeof(data), 44100);
/* Check if an error occured, and clean up if so. */
err = alGetError();
if(err != AL_NO_ERROR)
{
fprintf(stderr, "OpenAL Error: %s\n", alGetString(err));
if(alIsBuffer(buffer))
alDeleteBuffers(1, &buffer);
return 0;
}
return buffer;
}
int main(int argc, char *argv[])
{
PlaybackInfo playback = { NULL, NULL, 0 };
SDL_AudioSpec desired, obtained;
ALuint source, buffer;
ALCint attrs[16];
ALenum state;
(void)argc;
(void)argv;
/* Print out error if extension is missing. */
if(!alcIsExtensionPresent(NULL, "ALC_SOFT_loopback"))
{
fprintf(stderr, "Error: ALC_SOFT_loopback not supported!\n");
return 1;
}
/* Define a macro to help load the function pointers. */
#define LOAD_PROC(x) ((x) = alcGetProcAddress(NULL, #x))
LOAD_PROC(alcLoopbackOpenDeviceSOFT);
LOAD_PROC(alcIsRenderFormatSupportedSOFT);
LOAD_PROC(alcRenderSamplesSOFT);
#undef LOAD_PROC
if(SDL_Init(SDL_INIT_AUDIO) == -1)
{
fprintf(stderr, "Failed to init SDL audio: %s\n", SDL_GetError());
return 1;
}
/* Set up SDL audio with our requested format and callback. */
desired.channels = 2;
desired.format = AUDIO_S16SYS;
desired.freq = 44100;
desired.padding = 0;
desired.samples = 4096;
desired.callback = RenderSDLSamples;
desired.userdata = &playback;
if(SDL_OpenAudio(&desired, &obtained) != 0)
{
SDL_Quit();
fprintf(stderr, "Failed to open SDL audio: %s\n", SDL_GetError());
return 1;
}
/* Set up our OpenAL attributes based on what we got from SDL. */
attrs[0] = ALC_FORMAT_CHANNELS_SOFT;
if(obtained.channels == 1)
attrs[1] = ALC_MONO_SOFT;
else if(obtained.channels == 2)
attrs[1] = ALC_STEREO_SOFT;
else
{
fprintf(stderr, "Unhandled SDL channel count: %d\n", obtained.channels);
goto error;
}
attrs[2] = ALC_FORMAT_TYPE_SOFT;
if(obtained.format == AUDIO_U8)
attrs[3] = ALC_UNSIGNED_BYTE_SOFT;
else if(obtained.format == AUDIO_S8)
attrs[3] = ALC_BYTE_SOFT;
else if(obtained.format == AUDIO_U16SYS)
attrs[3] = ALC_UNSIGNED_SHORT_SOFT;
else if(obtained.format == AUDIO_S16SYS)
attrs[3] = ALC_SHORT_SOFT;
else
{
fprintf(stderr, "Unhandled SDL format: 0x%04x\n", obtained.format);
goto error;
}
attrs[4] = ALC_FREQUENCY;
attrs[5] = obtained.freq;
attrs[6] = 0; /* end of list */
playback.FrameSize = obtained.channels * SDL_AUDIO_BITSIZE(obtained.format) / 8;
/* Initialize OpenAL loopback device, using our format attributes. */
playback.Device = alcLoopbackOpenDeviceSOFT(NULL);
if(!playback.Device)
{
fprintf(stderr, "Failed to open loopback device!\n");
goto error;
}
/* Make sure the format is supported before setting them on the device. */
if(alcIsRenderFormatSupportedSOFT(playback.Device, attrs[5], attrs[1], attrs[3]) == ALC_FALSE)
{
fprintf(stderr, "Render format not supported: %s, %s, %dhz\n",
ChannelsName(attrs[1]), TypeName(attrs[3]), attrs[5]);
goto error;
}
playback.Context = alcCreateContext(playback.Device, attrs);
if(!playback.Context || alcMakeContextCurrent(playback.Context) == ALC_FALSE)
{
fprintf(stderr, "Failed to set an OpenAL audio context\n");
goto error;
}
/* Start SDL playing. Our callback (thus alcRenderSamplesSOFT) will now
* start being called regularly to update the AL playback state. */
SDL_PauseAudio(0);
/* Load the sound into a buffer. */
buffer = CreateSineWave();
if(!buffer)
{
SDL_CloseAudio();
alcDestroyContext(playback.Context);
alcCloseDevice(playback.Device);
SDL_Quit();
return 1;
}
/* Create the source to play the sound with. */
source = 0;
alGenSources(1, &source);
alSourcei(source, AL_BUFFER, buffer);
assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
/* Play the sound until it finishes. */
alSourcePlay(source);
do {
al_nssleep(10000000);
alGetSourcei(source, AL_SOURCE_STATE, &state);
} while(alGetError() == AL_NO_ERROR && state == AL_PLAYING);
/* All done. Delete resources, and close OpenAL. */
alDeleteSources(1, &source);
alDeleteBuffers(1, &buffer);
/* Stop SDL playing. */
SDL_PauseAudio(1);
/* Close up OpenAL and SDL. */
SDL_CloseAudio();
alcDestroyContext(playback.Context);
alcCloseDevice(playback.Device);
SDL_Quit();
return 0;
error:
SDL_CloseAudio();
if(playback.Context)
alcDestroyContext(playback.Context);
if(playback.Device)
alcCloseDevice(playback.Device);
SDL_Quit();
return 1;
}