mirror of
https://github.com/TorqueGameEngines/Torque3D.git
synced 2026-01-30 17:41:00 +00:00
a745fc3757
added libraries: opus flac libsndfile updated: libvorbis libogg openal - Everything works as expected for now. Bare in mind libsndfile needed the check for whether or not it could find the xiph libraries removed in order for this to work.
294 lines
8.5 KiB
C
294 lines
8.5 KiB
C
/*
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* OpenAL Loopback Example
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*
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* Copyright (c) 2013 by Chris Robinson <chris.kcat@gmail.com>
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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/* This file contains an example for using the loopback device for custom
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* output handling.
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*/
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#include <assert.h>
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#include <math.h>
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#include <stdio.h>
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#define SDL_MAIN_HANDLED
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#include "SDL.h"
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#include "SDL_audio.h"
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#include "SDL_error.h"
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#include "SDL_stdinc.h"
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#include "AL/al.h"
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#include "AL/alc.h"
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#include "AL/alext.h"
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#include "common/alhelpers.h"
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#ifndef SDL_AUDIO_MASK_BITSIZE
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#define SDL_AUDIO_MASK_BITSIZE (0xFF)
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#endif
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#ifndef SDL_AUDIO_BITSIZE
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#define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE)
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#endif
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#ifndef M_PI
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#define M_PI (3.14159265358979323846)
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#endif
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typedef struct {
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ALCdevice *Device;
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ALCcontext *Context;
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ALCsizei FrameSize;
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} PlaybackInfo;
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static LPALCLOOPBACKOPENDEVICESOFT alcLoopbackOpenDeviceSOFT;
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static LPALCISRENDERFORMATSUPPORTEDSOFT alcIsRenderFormatSupportedSOFT;
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static LPALCRENDERSAMPLESSOFT alcRenderSamplesSOFT;
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void SDLCALL RenderSDLSamples(void *userdata, Uint8 *stream, int len)
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{
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PlaybackInfo *playback = (PlaybackInfo*)userdata;
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alcRenderSamplesSOFT(playback->Device, stream, len/playback->FrameSize);
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}
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static const char *ChannelsName(ALCenum chans)
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{
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switch(chans)
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{
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case ALC_MONO_SOFT: return "Mono";
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case ALC_STEREO_SOFT: return "Stereo";
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case ALC_QUAD_SOFT: return "Quadraphonic";
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case ALC_5POINT1_SOFT: return "5.1 Surround";
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case ALC_6POINT1_SOFT: return "6.1 Surround";
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case ALC_7POINT1_SOFT: return "7.1 Surround";
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}
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return "Unknown Channels";
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}
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static const char *TypeName(ALCenum type)
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{
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switch(type)
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{
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case ALC_BYTE_SOFT: return "S8";
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case ALC_UNSIGNED_BYTE_SOFT: return "U8";
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case ALC_SHORT_SOFT: return "S16";
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case ALC_UNSIGNED_SHORT_SOFT: return "U16";
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case ALC_INT_SOFT: return "S32";
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case ALC_UNSIGNED_INT_SOFT: return "U32";
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case ALC_FLOAT_SOFT: return "Float32";
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}
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return "Unknown Type";
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}
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/* Creates a one second buffer containing a sine wave, and returns the new
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* buffer ID. */
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static ALuint CreateSineWave(void)
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{
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ALshort data[44100*4];
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ALuint buffer;
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ALenum err;
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ALuint i;
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for(i = 0;i < 44100*4;i++)
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data[i] = (ALshort)(sin(i/44100.0 * 1000.0 * 2.0*M_PI) * 32767.0);
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/* Buffer the audio data into a new buffer object. */
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buffer = 0;
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alGenBuffers(1, &buffer);
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alBufferData(buffer, AL_FORMAT_MONO16, data, sizeof(data), 44100);
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/* Check if an error occured, and clean up if so. */
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err = alGetError();
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if(err != AL_NO_ERROR)
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{
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fprintf(stderr, "OpenAL Error: %s\n", alGetString(err));
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if(alIsBuffer(buffer))
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alDeleteBuffers(1, &buffer);
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return 0;
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}
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return buffer;
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}
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int main(int argc, char *argv[])
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{
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PlaybackInfo playback = { NULL, NULL, 0 };
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SDL_AudioSpec desired, obtained;
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ALuint source, buffer;
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ALCint attrs[16];
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ALenum state;
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(void)argc;
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(void)argv;
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SDL_SetMainReady();
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/* Print out error if extension is missing. */
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if(!alcIsExtensionPresent(NULL, "ALC_SOFT_loopback"))
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{
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fprintf(stderr, "Error: ALC_SOFT_loopback not supported!\n");
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return 1;
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}
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/* Define a macro to help load the function pointers. */
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#define LOAD_PROC(T, x) ((x) = FUNCTION_CAST(T, alcGetProcAddress(NULL, #x)))
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LOAD_PROC(LPALCLOOPBACKOPENDEVICESOFT, alcLoopbackOpenDeviceSOFT);
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LOAD_PROC(LPALCISRENDERFORMATSUPPORTEDSOFT, alcIsRenderFormatSupportedSOFT);
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LOAD_PROC(LPALCRENDERSAMPLESSOFT, alcRenderSamplesSOFT);
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#undef LOAD_PROC
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if(SDL_Init(SDL_INIT_AUDIO) == -1)
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{
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fprintf(stderr, "Failed to init SDL audio: %s\n", SDL_GetError());
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return 1;
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}
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/* Set up SDL audio with our requested format and callback. */
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desired.channels = 2;
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desired.format = AUDIO_S16SYS;
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desired.freq = 44100;
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desired.padding = 0;
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desired.samples = 4096;
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desired.callback = RenderSDLSamples;
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desired.userdata = &playback;
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if(SDL_OpenAudio(&desired, &obtained) != 0)
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{
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SDL_Quit();
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fprintf(stderr, "Failed to open SDL audio: %s\n", SDL_GetError());
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return 1;
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}
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/* Set up our OpenAL attributes based on what we got from SDL. */
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attrs[0] = ALC_FORMAT_CHANNELS_SOFT;
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if(obtained.channels == 1)
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attrs[1] = ALC_MONO_SOFT;
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else if(obtained.channels == 2)
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attrs[1] = ALC_STEREO_SOFT;
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else
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{
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fprintf(stderr, "Unhandled SDL channel count: %d\n", obtained.channels);
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goto error;
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}
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attrs[2] = ALC_FORMAT_TYPE_SOFT;
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if(obtained.format == AUDIO_U8)
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attrs[3] = ALC_UNSIGNED_BYTE_SOFT;
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else if(obtained.format == AUDIO_S8)
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attrs[3] = ALC_BYTE_SOFT;
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else if(obtained.format == AUDIO_U16SYS)
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attrs[3] = ALC_UNSIGNED_SHORT_SOFT;
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else if(obtained.format == AUDIO_S16SYS)
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attrs[3] = ALC_SHORT_SOFT;
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else if(obtained.format == AUDIO_S32SYS)
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attrs[3] = ALC_INT_SOFT;
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else if(obtained.format == AUDIO_F32SYS)
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attrs[3] = ALC_FLOAT_SOFT;
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else
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{
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fprintf(stderr, "Unhandled SDL format: 0x%04x\n", obtained.format);
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goto error;
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}
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attrs[4] = ALC_FREQUENCY;
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attrs[5] = obtained.freq;
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attrs[6] = 0; /* end of list */
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playback.FrameSize = obtained.channels * SDL_AUDIO_BITSIZE(obtained.format) / 8;
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/* Initialize OpenAL loopback device, using our format attributes. */
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playback.Device = alcLoopbackOpenDeviceSOFT(NULL);
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if(!playback.Device)
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{
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fprintf(stderr, "Failed to open loopback device!\n");
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goto error;
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}
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/* Make sure the format is supported before setting them on the device. */
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if(alcIsRenderFormatSupportedSOFT(playback.Device, attrs[5], attrs[1], attrs[3]) == ALC_FALSE)
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{
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fprintf(stderr, "Render format not supported: %s, %s, %dhz\n",
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ChannelsName(attrs[1]), TypeName(attrs[3]), attrs[5]);
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goto error;
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}
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playback.Context = alcCreateContext(playback.Device, attrs);
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if(!playback.Context || alcMakeContextCurrent(playback.Context) == ALC_FALSE)
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{
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fprintf(stderr, "Failed to set an OpenAL audio context\n");
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goto error;
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}
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/* Start SDL playing. Our callback (thus alcRenderSamplesSOFT) will now
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* start being called regularly to update the AL playback state. */
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SDL_PauseAudio(0);
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/* Load the sound into a buffer. */
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buffer = CreateSineWave();
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if(!buffer)
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{
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SDL_CloseAudio();
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alcDestroyContext(playback.Context);
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alcCloseDevice(playback.Device);
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SDL_Quit();
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return 1;
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}
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/* Create the source to play the sound with. */
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source = 0;
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alGenSources(1, &source);
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alSourcei(source, AL_BUFFER, (ALint)buffer);
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assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
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/* Play the sound until it finishes. */
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alSourcePlay(source);
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do {
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al_nssleep(10000000);
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alGetSourcei(source, AL_SOURCE_STATE, &state);
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} while(alGetError() == AL_NO_ERROR && state == AL_PLAYING);
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/* All done. Delete resources, and close OpenAL. */
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alDeleteSources(1, &source);
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alDeleteBuffers(1, &buffer);
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/* Stop SDL playing. */
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SDL_PauseAudio(1);
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/* Close up OpenAL and SDL. */
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SDL_CloseAudio();
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alcDestroyContext(playback.Context);
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alcCloseDevice(playback.Device);
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SDL_Quit();
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return 0;
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error:
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SDL_CloseAudio();
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if(playback.Context)
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alcDestroyContext(playback.Context);
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if(playback.Device)
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alcCloseDevice(playback.Device);
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SDL_Quit();
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return 1;
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}
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