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- /*
- * 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;
- }
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