diff options
Diffstat (limited to 'examples')
| -rw-r--r-- | examples/alhrtf.c | 75 | ||||
| -rw-r--r-- | examples/allatency.c | 87 | ||||
| -rw-r--r-- | examples/alloopback.c | 31 | ||||
| -rw-r--r-- | examples/alreverb.c | 91 | ||||
| -rw-r--r-- | examples/alstream.c | 118 | ||||
| -rw-r--r-- | examples/common/alhelpers.c | 243 | ||||
| -rw-r--r-- | examples/common/alhelpers.h | 24 | ||||
| -rw-r--r-- | examples/common/sdl_sound.c | 164 | ||||
| -rw-r--r-- | examples/common/sdl_sound.h | 43 |
9 files changed, 252 insertions, 624 deletions
diff --git a/examples/alhrtf.c b/examples/alhrtf.c index 3964a7c6..f9150ae1 100644 --- a/examples/alhrtf.c +++ b/examples/alhrtf.c @@ -28,12 +28,13 @@ #include <assert.h> #include <math.h> +#include <SDL_sound.h> + #include "AL/al.h" #include "AL/alc.h" #include "AL/alext.h" #include "common/alhelpers.h" -#include "common/sdl_sound.h" #ifndef M_PI @@ -44,47 +45,63 @@ static LPALCGETSTRINGISOFT alcGetStringiSOFT; static LPALCRESETDEVICESOFT alcResetDeviceSOFT; /* LoadBuffer loads the named audio file into an OpenAL buffer object, and - * returns the new buffer ID. */ + * returns the new buffer ID. + */ static ALuint LoadSound(const char *filename) { - ALenum err, format, type, channels; - ALuint rate, buffer; - size_t datalen; - void *data; - FilePtr sound; + Sound_Sample *sample; + ALenum err, format; + ALuint buffer; + Uint32 slen; /* Open the audio file */ - sound = openAudioFile(filename, 1000); - if(!sound) + sample = Sound_NewSampleFromFile(filename, NULL, 65536); + if(!sample) { fprintf(stderr, "Could not open audio in %s\n", filename); - closeAudioFile(sound); return 0; } /* Get the sound format, and figure out the OpenAL format */ - if(getAudioInfo(sound, &rate, &channels, &type) != 0) + if(sample->actual.channels == 1) { - fprintf(stderr, "Error getting audio info for %s\n", filename); - closeAudioFile(sound); - return 0; + if(sample->actual.format == AUDIO_U8) + format = AL_FORMAT_MONO8; + else if(sample->actual.format == AUDIO_S16SYS) + format = AL_FORMAT_MONO16; + else + { + fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format); + Sound_FreeSample(sample); + return 0; + } } - - format = GetFormat(channels, type, NULL); - if(format == AL_NONE) + else if(sample->actual.channels == 2) { - fprintf(stderr, "Unsupported format (%s, %s) for %s\n", - ChannelsName(channels), TypeName(type), filename); - closeAudioFile(sound); + if(sample->actual.format == AUDIO_U8) + format = AL_FORMAT_STEREO8; + else if(sample->actual.format == AUDIO_S16SYS) + format = AL_FORMAT_STEREO16; + else + { + fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format); + Sound_FreeSample(sample); + return 0; + } + } + else + { + fprintf(stderr, "Unsupported channel count: %d\n", sample->actual.channels); + Sound_FreeSample(sample); return 0; } /* Decode the whole audio stream to a buffer. */ - data = decodeAudioStream(sound, &datalen); - if(!data) + slen = Sound_DecodeAll(sample); + if(!sample->buffer || slen == 0) { fprintf(stderr, "Failed to read audio from %s\n", filename); - closeAudioFile(sound); + Sound_FreeSample(sample); return 0; } @@ -92,9 +109,8 @@ static ALuint LoadSound(const char *filename) * close the file. */ buffer = 0; alGenBuffers(1, &buffer); - alBufferData(buffer, format, data, datalen, rate); - free(data); - closeAudioFile(sound); + alBufferData(buffer, format, sample->buffer, slen, sample->actual.rate); + Sound_FreeSample(sample); /* Check if an error occured, and clean up if so. */ err = alGetError(); @@ -219,10 +235,14 @@ int main(int argc, char **argv) } fflush(stdout); + /* Initialize SDL_sound. */ + Sound_Init(); + /* Load the sound into a buffer. */ buffer = LoadSound(soundname); if(!buffer) { + Sound_Quit(); CloseAL(); return 1; } @@ -263,10 +283,11 @@ int main(int argc, char **argv) alGetSourcei(source, AL_SOURCE_STATE, &state); } while(alGetError() == AL_NO_ERROR && state == AL_PLAYING); - /* All done. Delete resources, and close OpenAL. */ + /* All done. Delete resources, and close down SDL_sound and OpenAL. */ alDeleteSources(1, &source); alDeleteBuffers(1, &buffer); + Sound_Quit(); CloseAL(); return 0; diff --git a/examples/allatency.c b/examples/allatency.c index 56d96b9e..d561373f 100644 --- a/examples/allatency.c +++ b/examples/allatency.c @@ -27,16 +27,14 @@ #include <stdio.h> #include <assert.h> +#include <SDL_sound.h> + #include "AL/al.h" #include "AL/alc.h" #include "AL/alext.h" #include "common/alhelpers.h" -#include "common/sdl_sound.h" - -static LPALBUFFERSAMPLESSOFT alBufferSamplesSOFT = wrap_BufferSamples; -static LPALISBUFFERFORMATSUPPORTEDSOFT alIsBufferFormatSupportedSOFT; static LPALSOURCEDSOFT alSourcedSOFT; static LPALSOURCE3DSOFT alSource3dSOFT; @@ -52,47 +50,63 @@ static LPALGETSOURCE3I64SOFT alGetSource3i64SOFT; static LPALGETSOURCEI64VSOFT alGetSourcei64vSOFT; /* LoadBuffer loads the named audio file into an OpenAL buffer object, and - * returns the new buffer ID. */ + * returns the new buffer ID. + */ static ALuint LoadSound(const char *filename) { - ALenum err, format, type, channels; - ALuint rate, buffer; - size_t datalen; - void *data; - FilePtr sound; + Sound_Sample *sample; + ALenum err, format; + ALuint buffer; + Uint32 slen; /* Open the audio file */ - sound = openAudioFile(filename, 1000); - if(!sound) + sample = Sound_NewSampleFromFile(filename, NULL, 65536); + if(!sample) { fprintf(stderr, "Could not open audio in %s\n", filename); - closeAudioFile(sound); return 0; } /* Get the sound format, and figure out the OpenAL format */ - if(getAudioInfo(sound, &rate, &channels, &type) != 0) + if(sample->actual.channels == 1) { - fprintf(stderr, "Error getting audio info for %s\n", filename); - closeAudioFile(sound); - return 0; + if(sample->actual.format == AUDIO_U8) + format = AL_FORMAT_MONO8; + else if(sample->actual.format == AUDIO_S16SYS) + format = AL_FORMAT_MONO16; + else + { + fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format); + Sound_FreeSample(sample); + return 0; + } } - - format = GetFormat(channels, type, alIsBufferFormatSupportedSOFT); - if(format == AL_NONE) + else if(sample->actual.channels == 2) { - fprintf(stderr, "Unsupported format (%s, %s) for %s\n", - ChannelsName(channels), TypeName(type), filename); - closeAudioFile(sound); + if(sample->actual.format == AUDIO_U8) + format = AL_FORMAT_STEREO8; + else if(sample->actual.format == AUDIO_S16SYS) + format = AL_FORMAT_STEREO16; + else + { + fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format); + Sound_FreeSample(sample); + return 0; + } + } + else + { + fprintf(stderr, "Unsupported channel count: %d\n", sample->actual.channels); + Sound_FreeSample(sample); return 0; } /* Decode the whole audio stream to a buffer. */ - data = decodeAudioStream(sound, &datalen); - if(!data) + slen = Sound_DecodeAll(sample); + if(!sample->buffer || slen == 0) { fprintf(stderr, "Failed to read audio from %s\n", filename); - closeAudioFile(sound); + Sound_FreeSample(sample); return 0; } @@ -100,17 +114,15 @@ static ALuint LoadSound(const char *filename) * close the file. */ buffer = 0; alGenBuffers(1, &buffer); - alBufferSamplesSOFT(buffer, rate, format, BytesToFrames(datalen, channels, type), - channels, type, data); - free(data); - closeAudioFile(sound); + alBufferData(buffer, format, sample->buffer, slen, sample->actual.rate); + Sound_FreeSample(sample); /* 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)) + if(buffer && alIsBuffer(buffer)) alDeleteBuffers(1, &buffer); return 0; } @@ -158,18 +170,16 @@ int main(int argc, char **argv) LOAD_PROC(alGetSourcei64SOFT); LOAD_PROC(alGetSource3i64SOFT); LOAD_PROC(alGetSourcei64vSOFT); - - if(alIsExtensionPresent("AL_SOFT_buffer_samples")) - { - LOAD_PROC(alBufferSamplesSOFT); - LOAD_PROC(alIsBufferFormatSupportedSOFT); - } #undef LOAD_PROC + /* Initialize SDL_sound. */ + Sound_Init(); + /* Load the sound into a buffer. */ buffer = LoadSound(argv[0]); if(!buffer) { + Sound_Quit(); CloseAL(); return 1; } @@ -195,10 +205,11 @@ int main(int argc, char **argv) } while(alGetError() == AL_NO_ERROR && state == AL_PLAYING); printf("\n"); - /* All done. Delete resources, and close OpenAL. */ + /* All done. Delete resources, and close down SDL_sound and OpenAL. */ alDeleteSources(1, &source); alDeleteBuffers(1, &buffer); + Sound_Quit(); CloseAL(); return 0; diff --git a/examples/alloopback.c b/examples/alloopback.c index c5dee36d..95ac433f 100644 --- a/examples/alloopback.c +++ b/examples/alloopback.c @@ -61,6 +61,35 @@ void SDLCALL RenderSDLSamples(void *userdata, Uint8 *stream, int len) } +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) @@ -169,7 +198,7 @@ int main(int argc, char *argv[]) attrs[6] = 0; /* end of list */ - playback.FrameSize = FramesToBytes(1, attrs[1], attrs[3]); + playback.FrameSize = obtained.channels * SDL_AUDIO_BITSIZE(obtained.format) / 8; /* Initialize OpenAL loopback device, using our format attributes. */ playback.Device = alcLoopbackOpenDeviceSOFT(NULL); diff --git a/examples/alreverb.c b/examples/alreverb.c index ec71f354..e6c9e606 100644 --- a/examples/alreverb.c +++ b/examples/alreverb.c @@ -27,17 +27,15 @@ #include <stdio.h> #include <assert.h> +#include <SDL_sound.h> + #include "AL/al.h" #include "AL/alc.h" #include "AL/alext.h" #include "AL/efx-presets.h" #include "common/alhelpers.h" -#include "common/sdl_sound.h" - -static LPALBUFFERSAMPLESSOFT alBufferSamplesSOFT = wrap_BufferSamples; -static LPALISBUFFERFORMATSUPPORTEDSOFT alIsBufferFormatSupportedSOFT; /* Effect object functions */ static LPALGENEFFECTS alGenEffects; @@ -145,46 +143,63 @@ static ALuint LoadEffect(const EFXEAXREVERBPROPERTIES *reverb) /* LoadBuffer loads the named audio file into an OpenAL buffer object, and - * returns the new buffer ID. */ + * returns the new buffer ID. + */ static ALuint LoadSound(const char *filename) { - ALenum err, format, type, channels; - ALuint rate, buffer; - size_t datalen; - void *data; - FilePtr sound; - - /* Open the file and get the first stream from it */ - sound = openAudioFile(filename, 1000); - if(!sound) + Sound_Sample *sample; + ALenum err, format; + ALuint buffer; + Uint32 slen; + + /* Open the audio file */ + sample = Sound_NewSampleFromFile(filename, NULL, 65536); + if(!sample) { fprintf(stderr, "Could not open audio in %s\n", filename); return 0; } /* Get the sound format, and figure out the OpenAL format */ - if(getAudioInfo(sound, &rate, &channels, &type) != 0) + if(sample->actual.channels == 1) { - fprintf(stderr, "Error getting audio info for %s\n", filename); - closeAudioFile(sound); - return 0; + if(sample->actual.format == AUDIO_U8) + format = AL_FORMAT_MONO8; + else if(sample->actual.format == AUDIO_S16SYS) + format = AL_FORMAT_MONO16; + else + { + fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format); + Sound_FreeSample(sample); + return 0; + } } - - format = GetFormat(channels, type, alIsBufferFormatSupportedSOFT); - if(format == AL_NONE) + else if(sample->actual.channels == 2) + { + if(sample->actual.format == AUDIO_U8) + format = AL_FORMAT_STEREO8; + else if(sample->actual.format == AUDIO_S16SYS) + format = AL_FORMAT_STEREO16; + else + { + fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format); + Sound_FreeSample(sample); + return 0; + } + } + else { - fprintf(stderr, "Unsupported format (%s, %s) for %s\n", - ChannelsName(channels), TypeName(type), filename); - closeAudioFile(sound); + fprintf(stderr, "Unsupported channel count: %d\n", sample->actual.channels); + Sound_FreeSample(sample); return 0; } /* Decode the whole audio stream to a buffer. */ - data = decodeAudioStream(sound, &datalen); - if(!data) + slen = Sound_DecodeAll(sample); + if(!sample->buffer || slen == 0) { fprintf(stderr, "Failed to read audio from %s\n", filename); - closeAudioFile(sound); + Sound_FreeSample(sample); return 0; } @@ -192,17 +207,15 @@ static ALuint LoadSound(const char *filename) * close the file. */ buffer = 0; alGenBuffers(1, &buffer); - alBufferSamplesSOFT(buffer, rate, format, BytesToFrames(datalen, channels, type), - channels, type, data); - free(data); - closeAudioFile(sound); + alBufferData(buffer, format, sample->buffer, slen, sample->actual.rate); + Sound_FreeSample(sample); /* 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)) + if(buffer && alIsBuffer(buffer)) alDeleteBuffers(1, &buffer); return 0; } @@ -261,19 +274,17 @@ int main(int argc, char **argv) LOAD_PROC(alGetAuxiliaryEffectSlotiv); LOAD_PROC(alGetAuxiliaryEffectSlotf); LOAD_PROC(alGetAuxiliaryEffectSlotfv); - - if(alIsExtensionPresent("AL_SOFT_buffer_samples")) - { - LOAD_PROC(alBufferSamplesSOFT); - LOAD_PROC(alIsBufferFormatSupportedSOFT); - } #undef LOAD_PROC + /* Initialize SDL_sound. */ + Sound_Init(); + /* Load the sound into a buffer. */ buffer = LoadSound(argv[0]); if(!buffer) { CloseAL(); + Sound_Quit(); return 1; } @@ -282,6 +293,7 @@ int main(int argc, char **argv) if(!effect) { alDeleteBuffers(1, &buffer); + Sound_Quit(); CloseAL(); return 1; } @@ -316,12 +328,13 @@ int main(int argc, char **argv) alGetSourcei(source, AL_SOURCE_STATE, &state); } while(alGetError() == AL_NO_ERROR && state == AL_PLAYING); - /* All done. Delete resources, and close OpenAL. */ + /* All done. Delete resources, and close down SDL_sound and OpenAL. */ alDeleteSources(1, &source); alDeleteAuxiliaryEffectSlots(1, &slot); alDeleteEffects(1, &effect); alDeleteBuffers(1, &buffer); + Sound_Quit(); CloseAL(); return 0; diff --git a/examples/alstream.c b/examples/alstream.c index 65a04475..d13899d0 100644 --- a/examples/alstream.c +++ b/examples/alstream.c @@ -30,16 +30,13 @@ #include <signal.h> #include <assert.h> +#include <SDL_sound.h> + #include "AL/al.h" #include "AL/alc.h" #include "AL/alext.h" #include "common/alhelpers.h" -#include "common/sdl_sound.h" - - -static LPALBUFFERSAMPLESSOFT alBufferSamplesSOFT = wrap_BufferSamples; -static LPALISBUFFERFORMATSUPPORTEDSOFT alIsBufferFormatSupportedSOFT; /* Define the number of buffers and buffer size (in milliseconds) to use. 4 @@ -54,13 +51,11 @@ typedef struct StreamPlayer { ALuint source; /* Handle for the audio file */ - FilePtr file; + Sound_Sample *sample; /* The format of the output stream */ ALenum format; - ALenum channels; - ALenum type; - ALuint rate; + ALsizei srate; } StreamPlayer; static StreamPlayer *NewPlayer(void); @@ -77,11 +72,9 @@ static StreamPlayer *NewPlayer(void) { StreamPlayer *player; - player = malloc(sizeof(*player)); + player = calloc(1, sizeof(*player)); assert(player != NULL); - memset(player, 0, sizeof(*player)); - /* Generate the buffers and source */ alGenBuffers(NUM_BUFFERS, player->buffers); assert(alGetError() == AL_NO_ERROR && "Could not create buffers"); @@ -119,37 +112,63 @@ static void DeletePlayer(StreamPlayer *player) * it will be closed first. */ static int OpenPlayerFile(StreamPlayer *player, const char *filename) { + Uint32 frame_size; + ClosePlayerFile(player); /* Open the file and get the first stream from it */ - player->file = openAudioFile(filename, BUFFER_TIME_MS); - if(!player->file) + player->sample = Sound_NewSampleFromFile(filename, NULL, 0); + if(!player->sample) { fprintf(stderr, "Could not open audio in %s\n", filename); goto error; } /* Get the stream format, and figure out the OpenAL format */ - if(getAudioInfo(player->file, &player->rate, &player->channels, &player->type) != 0) + if(player->sample->actual.channels == 1) { - fprintf(stderr, "Error getting audio info for %s\n", filename); - goto error; + if(player->sample->actual.format == AUDIO_U8) + player->format = AL_FORMAT_MONO8; + else if(player->sample->actual.format == AUDIO_S16SYS) + player->format = AL_FORMAT_MONO16; + else + { + fprintf(stderr, "Unsupported sample format: 0x%04x\n", player->sample->actual.format); + goto error; + } } - - player->format = GetFormat(player->channels, player->type, alIsBufferFormatSupportedSOFT); - if(player->format == 0) + else if(player->sample->actual.channels == 2) { - fprintf(stderr, "Unsupported format (%s, %s) for %s\n", - ChannelsName(player->channels), TypeName(player->type), - filename); + if(player->sample->actual.format == AUDIO_U8) + player->format = AL_FORMAT_STEREO8; + else if(player->sample->actual.format == AUDIO_S16SYS) + player->format = AL_FORMAT_STEREO16; + else + { + fprintf(stderr, "Unsupported sample format: 0x%04x\n", player->sample->actual.format); + goto error; + } + } + else + { + fprintf(stderr, "Unsupported channel count: %d\n", player->sample->actual.channels); goto error; } + player->srate = player->sample->actual.rate; + + frame_size = player->sample->actual.channels * + SDL_AUDIO_BITSIZE(player->sample->actual.format) / 8; + + /* Set the buffer size, given the desired millisecond length. */ + Sound_SetBufferSize(player->sample, (Uint32)((Uint64)player->srate*BUFFER_TIME_MS/1000) * + frame_size); return 1; error: - closeAudioFile(player->file); - player->file = NULL; + if(player->sample) + Sound_FreeSample(player->sample); + player->sample = NULL; return 0; } @@ -157,8 +176,9 @@ error: /* Closes the audio file stream */ static void ClosePlayerFile(StreamPlayer *player) { - closeAudioFile(player->file); - player->file = NULL; + if(player->sample) + Sound_FreeSample(player->sample); + player->sample = NULL; } @@ -174,16 +194,12 @@ static int StartPlayer(StreamPlayer *player) /* Fill the buffer queue */ for(i = 0;i < NUM_BUFFERS;i++) { - uint8_t *data; - size_t got; - /* Get some data to give it to the buffer */ - data = getAudioData(player->file, &got); - if(!data) break; + Uint32 slen = Sound_Decode(player->sample); + if(slen == 0) break; - alBufferSamplesSOFT(player->buffers[i], player->rate, player->format, - BytesToFrames(got, player->channels, player->type), - player->channels, player->type, data); + alBufferData(player->buffers[i], player->format, + player->sample->buffer, slen, player->srate); } if(alGetError() != AL_NO_ERROR) { @@ -220,20 +236,21 @@ static int UpdatePlayer(StreamPlayer *player) while(processed > 0) { ALuint bufid; - uint8_t *data; - size_t got; + Uint32 slen; alSourceUnqueueBuffers(player->source, 1, &bufid); processed--; + if((player->sample->flags&(SOUND_SAMPLEFLAG_EOF|SOUND_SAMPLEFLAG_ERROR))) + continue; + /* Read the next chunk of data, refill the buffer, and queue it * back on the source */ - data = getAudioData(player->file, &got); - if(data != NULL) + slen = Sound_Decode(player->sample); + if(slen > 0) { - alBufferSamplesSOFT(bufid, player->rate, player->format, - BytesToFrames(got, player->channels, player->type), - player->channels, player->type, data); + alBufferData(bufid, player->format, player->sample->buffer, slen, + player->srate); alSourceQueueBuffers(player->source, 1, &bufid); } if(alGetError() != AL_NO_ERROR) @@ -281,14 +298,7 @@ int main(int argc, char **argv) if(InitAL(&argv, &argc) != 0) return 1; - if(alIsExtensionPresent("AL_SOFT_buffer_samples")) - { - printf("AL_SOFT_buffer_samples supported!\n"); - alBufferSamplesSOFT = alGetProcAddress("alBufferSamplesSOFT"); - alIsBufferFormatSupportedSOFT = alGetProcAddress("alIsBufferFormatSupportedSOFT"); - } - else - printf("AL_SOFT_buffer_samples not supported\n"); + Sound_Init(); player = NewPlayer(); @@ -307,9 +317,8 @@ int main(int argc, char **argv) else namepart = argv[i]; - printf("Playing: %s (%s, %s, %dhz)\n", namepart, - TypeName(player->type), ChannelsName(player->channels), - player->rate); + printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->format), + player->srate); fflush(stdout); if(!StartPlayer(player)) @@ -326,10 +335,11 @@ int main(int argc, char **argv) } printf("Done.\n"); - /* All files done. Delete the player, and close OpenAL */ + /* All files done. Delete the player, and close down SDL_sound and OpenAL */ DeletePlayer(player); player = NULL; + Sound_Quit(); CloseAL(); return 0; diff --git a/examples/common/alhelpers.c b/examples/common/alhelpers.c index 43548b5c..fab039e9 100644 --- a/examples/common/alhelpers.c +++ b/examples/common/alhelpers.c @@ -103,243 +103,14 @@ void CloseAL(void) } -/* GetFormat retrieves a compatible buffer format given the channel config and - * sample type. If an alIsBufferFormatSupportedSOFT-compatible function is - * provided, it will be called to find the closest-matching format from - * AL_SOFT_buffer_samples. Returns AL_NONE (0) if no supported format can be - * found. */ -ALenum GetFormat(ALenum channels, ALenum type, LPALISBUFFERFORMATSUPPORTEDSOFT palIsBufferFormatSupportedSOFT) +const char *FormatName(ALenum format) { - ALenum format = AL_NONE; - - /* If using AL_SOFT_buffer_samples, try looking through its formats */ - if(palIsBufferFormatSupportedSOFT) - { - /* AL_SOFT_buffer_samples is more lenient with matching formats. The - * specified sample type does not need to match the returned format, - * but it is nice to try to get something close. */ - if(type == AL_UNSIGNED_BYTE_SOFT || type == AL_BYTE_SOFT) - { - if(channels == AL_MONO_SOFT) format = AL_MONO8_SOFT; - else if(channels == AL_STEREO_SOFT) format = AL_STEREO8_SOFT; - else if(channels == AL_QUAD_SOFT) format = AL_QUAD8_SOFT; - else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_8_SOFT; - else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_8_SOFT; - else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_8_SOFT; - } - else if(type == AL_UNSIGNED_SHORT_SOFT || type == AL_SHORT_SOFT) - { - if(channels == AL_MONO_SOFT) format = AL_MONO16_SOFT; - else if(channels == AL_STEREO_SOFT) format = AL_STEREO16_SOFT; - else if(channels == AL_QUAD_SOFT) format = AL_QUAD16_SOFT; - else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_16_SOFT; - else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_16_SOFT; - else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_16_SOFT; - } - else if(type == AL_UNSIGNED_BYTE3_SOFT || type == AL_BYTE3_SOFT || - type == AL_UNSIGNED_INT_SOFT || type == AL_INT_SOFT || - type == AL_FLOAT_SOFT || type == AL_DOUBLE_SOFT) - { - if(channels == AL_MONO_SOFT) format = AL_MONO32F_SOFT; - else if(channels == AL_STEREO_SOFT) format = AL_STEREO32F_SOFT; - else if(channels == AL_QUAD_SOFT) format = AL_QUAD32F_SOFT; - else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_32F_SOFT; - else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_32F_SOFT; - else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_32F_SOFT; - } - - if(format != AL_NONE && !palIsBufferFormatSupportedSOFT(format)) - format = AL_NONE; - - /* A matching format was not found or supported. Try 32-bit float. */ - if(format == AL_NONE) - { - if(channels == AL_MONO_SOFT) format = AL_MONO32F_SOFT; - else if(channels == AL_STEREO_SOFT) format = AL_STEREO32F_SOFT; - else if(channels == AL_QUAD_SOFT) format = AL_QUAD32F_SOFT; - else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_32F_SOFT; - else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_32F_SOFT; - else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_32F_SOFT; - - if(format != AL_NONE && !palIsBufferFormatSupportedSOFT(format)) - format = AL_NONE; - } - /* 32-bit float not supported. Try 16-bit int. */ - if(format == AL_NONE) - { - if(channels == AL_MONO_SOFT) format = AL_MONO16_SOFT; - else if(channels == AL_STEREO_SOFT) format = AL_STEREO16_SOFT; - else if(channels == AL_QUAD_SOFT) format = AL_QUAD16_SOFT; - else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_16_SOFT; - else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_16_SOFT; - else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_16_SOFT; - - if(format != AL_NONE && !palIsBufferFormatSupportedSOFT(format)) - format = AL_NONE; - } - /* 16-bit int not supported. Try 8-bit int. */ - if(format == AL_NONE) - { - if(channels == AL_MONO_SOFT) format = AL_MONO8_SOFT; - else if(channels == AL_STEREO_SOFT) format = AL_STEREO8_SOFT; - else if(channels == AL_QUAD_SOFT) format = AL_QUAD8_SOFT; - else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_8_SOFT; - else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_8_SOFT; - else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_8_SOFT; - - if(format != AL_NONE && !palIsBufferFormatSupportedSOFT(format)) - format = AL_NONE; - } - - return format; - } - - /* We use the AL_EXT_MCFORMATS extension to provide output of Quad, 5.1, - * and 7.1 channel configs, AL_EXT_FLOAT32 for 32-bit float samples, and - * AL_EXT_DOUBLE for 64-bit float samples. */ - if(type == AL_UNSIGNED_BYTE_SOFT) - { - if(channels == AL_MONO_SOFT) - format = AL_FORMAT_MONO8; - else if(channels == AL_STEREO_SOFT) - format = AL_FORMAT_STEREO8; - else if(alIsExtensionPresent("AL_EXT_MCFORMATS")) - { - if(channels == AL_QUAD_SOFT) - format = alGetEnumValue("AL_FORMAT_QUAD8"); - else if(channels == AL_5POINT1_SOFT) - format = alGetEnumValue("AL_FORMAT_51CHN8"); - else if(channels == AL_6POINT1_SOFT) - format = alGetEnumValue("AL_FORMAT_61CHN8"); - else if(channels == AL_7POINT1_SOFT) - format = alGetEnumValue("AL_FORMAT_71CHN8"); - } - } - else if(type == AL_SHORT_SOFT) - { - if(channels == AL_MONO_SOFT) - format = AL_FORMAT_MONO16; - else if(channels == AL_STEREO_SOFT) - format = AL_FORMAT_STEREO16; - else if(alIsExtensionPresent("AL_EXT_MCFORMATS")) - { - if(channels == AL_QUAD_SOFT) - format = alGetEnumValue("AL_FORMAT_QUAD16"); - else if(channels == AL_5POINT1_SOFT) - format = alGetEnumValue("AL_FORMAT_51CHN16"); - else if(channels == AL_6POINT1_SOFT) - format = alGetEnumValue("AL_FORMAT_61CHN16"); - else if(channels == AL_7POINT1_SOFT) - format = alGetEnumValue("AL_FORMAT_71CHN16"); - } - } - else if(type == AL_FLOAT_SOFT && alIsExtensionPresent("AL_EXT_FLOAT32")) - { - if(channels == AL_MONO_SOFT) - format = alGetEnumValue("AL_FORMAT_MONO_FLOAT32"); - else if(channels == AL_STEREO_SOFT) - format = alGetEnumValue("AL_FORMAT_STEREO_FLOAT32"); - else if(alIsExtensionPresent("AL_EXT_MCFORMATS")) - { - if(channels == AL_QUAD_SOFT) - format = alGetEnumValue("AL_FORMAT_QUAD32"); - else if(channels == AL_5POINT1_SOFT) - format = alGetEnumValue("AL_FORMAT_51CHN32"); - else if(channels == AL_6POINT1_SOFT) - format = alGetEnumValue("AL_FORMAT_61CHN32"); - else if(channels == AL_7POINT1_SOFT) - format = alGetEnumValue("AL_FORMAT_71CHN32"); - } - } - else if(type == AL_DOUBLE_SOFT && alIsExtensionPresent("AL_EXT_DOUBLE")) - { - if(channels == AL_MONO_SOFT) - format = alGetEnumValue("AL_FORMAT_MONO_DOUBLE"); - else if(channels == AL_STEREO_SOFT) - format = alGetEnumValue("AL_FORMAT_STEREO_DOUBLE"); - } - - /* NOTE: It seems OSX returns -1 from alGetEnumValue for unknown enums, as - * opposed to 0. Correct it. */ - if(format == -1) - format = 0; - - return format; -} - - -void AL_APIENTRY wrap_BufferSamples(ALuint buffer, ALuint samplerate, - ALenum internalformat, ALsizei samples, - ALenum channels, ALenum type, - const ALvoid *data) -{ - alBufferData(buffer, internalformat, data, - FramesToBytes(samples, channels, type), - samplerate); -} - - -const char *ChannelsName(ALenum chans) -{ - switch(chans) - { - case AL_MONO_SOFT: return "Mono"; - case AL_STEREO_SOFT: return "Stereo"; - case AL_REAR_SOFT: return "Rear"; - case AL_QUAD_SOFT: return "Quadraphonic"; - case AL_5POINT1_SOFT: return "5.1 Surround"; - case AL_6POINT1_SOFT: return "6.1 Surround"; - case AL_7POINT1_SOFT: return "7.1 Surround"; - } - return "Unknown Channels"; -} - -const char *TypeName(ALenum type) -{ - switch(type) + switch(format) { - case AL_BYTE_SOFT: return "S8"; - case AL_UNSIGNED_BYTE_SOFT: return "U8"; - case AL_SHORT_SOFT: return "S16"; - case AL_UNSIGNED_SHORT_SOFT: return "U16"; - case AL_INT_SOFT: return "S32"; - case AL_UNSIGNED_INT_SOFT: return "U32"; - case AL_FLOAT_SOFT: return "Float32"; - case AL_DOUBLE_SOFT: return "Float64"; + case AL_FORMAT_MONO8: return "Mono, U8"; + case AL_FORMAT_MONO16: return "Mono, S16"; + case AL_FORMAT_STEREO8: return "Stereo, U8"; + case AL_FORMAT_STEREO16: return "Stereo, S16"; } - return "Unknown Type"; -} - - -ALsizei FramesToBytes(ALsizei size, ALenum channels, ALenum type) -{ - switch(channels) - { - case AL_MONO_SOFT: size *= 1; break; - case AL_STEREO_SOFT: size *= 2; break; - case AL_REAR_SOFT: size *= 2; break; - case AL_QUAD_SOFT: size *= 4; break; - case AL_5POINT1_SOFT: size *= 6; break; - case AL_6POINT1_SOFT: size *= 7; break; - case AL_7POINT1_SOFT: size *= 8; break; - } - - switch(type) - { - case AL_BYTE_SOFT: size *= sizeof(ALbyte); break; - case AL_UNSIGNED_BYTE_SOFT: size *= sizeof(ALubyte); break; - case AL_SHORT_SOFT: size *= sizeof(ALshort); break; - case AL_UNSIGNED_SHORT_SOFT: size *= sizeof(ALushort); break; - case AL_INT_SOFT: size *= sizeof(ALint); break; - case AL_UNSIGNED_INT_SOFT: size *= sizeof(ALuint); break; - case AL_FLOAT_SOFT: size *= sizeof(ALfloat); break; - case AL_DOUBLE_SOFT: size *= sizeof(ALdouble); break; - } - - return size; -} - -ALsizei BytesToFrames(ALsizei size, ALenum channels, ALenum type) -{ - return size / FramesToBytes(1, channels, type); + return "Unknown Format"; } diff --git a/examples/common/alhelpers.h b/examples/common/alhelpers.h index 9f60df2a..41a7ce58 100644 --- a/examples/common/alhelpers.h +++ b/examples/common/alhelpers.h @@ -11,28 +11,8 @@ extern "C" { #endif /* __cplusplus */ -/* Some helper functions to get the name from the channel and type enums. */ -const char *ChannelsName(ALenum chans); -const char *TypeName(ALenum type); - -/* Helpers to convert frame counts and byte lengths. */ -ALsizei FramesToBytes(ALsizei size, ALenum channels, ALenum type); -ALsizei BytesToFrames(ALsizei size, ALenum channels, ALenum type); - -/* Retrieves a compatible buffer format given the channel configuration and - * sample type. If an alIsBufferFormatSupportedSOFT-compatible function is - * provided, it will be called to find the closest-matching format from - * AL_SOFT_buffer_samples. Returns AL_NONE (0) if no supported format can be - * found. */ -ALenum GetFormat(ALenum channels, ALenum type, LPALISBUFFERFORMATSUPPORTEDSOFT palIsBufferFormatSupportedSOFT); - -/* Loads samples into a buffer using the standard alBufferData call, but with a - * LPALBUFFERSAMPLESSOFT-compatible prototype. Assumes internalformat is valid - * for alBufferData, and that channels and type match it. */ -void AL_APIENTRY wrap_BufferSamples(ALuint buffer, ALuint samplerate, - ALenum internalformat, ALsizei samples, - ALenum channels, ALenum type, - const ALvoid *data); +/* Some helper functions to get the name from the format enums. */ +const char *FormatName(ALenum type); /* Easy device init/deinit functions. InitAL returns 0 on success. */ int InitAL(char ***argv, int *argc); diff --git a/examples/common/sdl_sound.c b/examples/common/sdl_sound.c deleted file mode 100644 index 79a5bf32..00000000 --- a/examples/common/sdl_sound.c +++ /dev/null @@ -1,164 +0,0 @@ -/* - * SDL_sound Decoder Helpers - * - * Copyright (c) 2013 by Chris Robinson <[email protected]> - * - * 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 routines for helping to decode audio using SDL_sound. - * There's very little OpenAL-specific code here. - */ -#include "sdl_sound.h" - -#include <string.h> -#include <stdlib.h> -#include <stdio.h> -#include <signal.h> -#include <assert.h> - -#include <SDL_sound.h> - -#include "AL/al.h" -#include "AL/alc.h" -#include "AL/alext.h" - -#include "alhelpers.h" - - -static int done_init = 0; - -FilePtr openAudioFile(const char *fname, size_t buftime_ms) -{ - FilePtr file; - ALuint rate; - Uint32 bufsize; - ALenum chans, type; - - /* We need to make sure SDL_sound is initialized. */ - if(!done_init) - { - Sound_Init(); - done_init = 1; - } - - file = Sound_NewSampleFromFile(fname, NULL, 0); - if(!file) - { - fprintf(stderr, "Failed to open %s: %s\n", fname, Sound_GetError()); - return NULL; - } - - if(getAudioInfo(file, &rate, &chans, &type) != 0) - { - Sound_FreeSample(file); - return NULL; - } - - bufsize = FramesToBytes((ALsizei)(buftime_ms/1000.0*rate), chans, type); - if(Sound_SetBufferSize(file, bufsize) == 0) - { - fprintf(stderr, "Failed to set buffer size to %u bytes: %s\n", bufsize, Sound_GetError()); - Sound_FreeSample(file); - return NULL; - } - - return file; -} - -void closeAudioFile(FilePtr file) -{ - if(file) - Sound_FreeSample(file); -} - - -int getAudioInfo(FilePtr file, ALuint *rate, ALenum *channels, ALenum *type) -{ - if(file->actual.channels == 1) - *channels = AL_MONO_SOFT; - else if(file->actual.channels == 2) - *channels = AL_STEREO_SOFT; - else - { - fprintf(stderr, "Unsupported channel count: %d\n", file->actual.channels); - return 1; - } - - if(file->actual.format == AUDIO_U8) - *type = AL_UNSIGNED_BYTE_SOFT; - else if(file->actual.format == AUDIO_S8) - *type = AL_BYTE_SOFT; - else if(file->actual.format == AUDIO_U16LSB || file->actual.format == AUDIO_U16MSB) - *type = AL_UNSIGNED_SHORT_SOFT; - else if(file->actual.format == AUDIO_S16LSB || file->actual.format == AUDIO_S16MSB) - *type = AL_SHORT_SOFT; - else - { - fprintf(stderr, "Unsupported sample format: 0x%04x\n", file->actual.format); - return 1; - } - - *rate = file->actual.rate; - - return 0; -} - - -uint8_t *getAudioData(FilePtr file, size_t *length) -{ - *length = Sound_Decode(file); - if(*length == 0) - return NULL; - if((file->actual.format == AUDIO_U16LSB && AUDIO_U16LSB != AUDIO_U16SYS) || - (file->actual.format == AUDIO_U16MSB && AUDIO_U16MSB != AUDIO_U16SYS) || - (file->actual.format == AUDIO_S16LSB && AUDIO_S16LSB != AUDIO_S16SYS) || - (file->actual.format == AUDIO_S16MSB && AUDIO_S16MSB != AUDIO_S16SYS)) - { - /* Swap bytes if the decoded endianness doesn't match the system. */ - char *buffer = file->buffer; - size_t i; - for(i = 0;i < *length;i+=2) - { - char b = buffer[i]; - buffer[i] = buffer[i+1]; - buffer[i+1] = b; - } - } - return file->buffer; -} - -void *decodeAudioStream(FilePtr file, size_t *length) -{ - Uint32 got; - char *mem; - - got = Sound_DecodeAll(file); - if(got == 0) - { - *length = 0; - return NULL; - } - - mem = malloc(got); - memcpy(mem, file->buffer, got); - - *length = got; - return mem; -} diff --git a/examples/common/sdl_sound.h b/examples/common/sdl_sound.h deleted file mode 100644 index e93ab92b..00000000 --- a/examples/common/sdl_sound.h +++ /dev/null @@ -1,43 +0,0 @@ -#ifndef EXAMPLES_SDL_SOUND_H -#define EXAMPLES_SDL_SOUND_H - -#include "AL/al.h" - -#include <SDL_sound.h> - -#ifdef __cplusplus -extern "C" { -#endif /* __cplusplus */ - -/* Opaque handles to files and streams. Apps don't need to concern themselves - * with the internals */ -typedef Sound_Sample *FilePtr; - -/* Opens a file with SDL_sound, and specifies the size of the sample buffer in - * milliseconds. */ -FilePtr openAudioFile(const char *fname, size_t buftime_ms); - -/* Closes/frees an opened file */ -void closeAudioFile(FilePtr file); - -/* Returns information about the given audio stream. Returns 0 on success. */ -int getAudioInfo(FilePtr file, ALuint *rate, ALenum *channels, ALenum *type); - -/* Returns a pointer to the next available chunk of decoded audio. The size (in - * bytes) of the returned data buffer is stored in 'length', and the returned - * pointer is only valid until the next call to getAudioData. */ -uint8_t *getAudioData(FilePtr file, size_t *length); - -/* Decodes all remaining data from the stream and returns a buffer containing - * the audio data, with the size stored in 'length'. The returned pointer must - * be freed with a call to free(). Note that since this decodes the whole - * stream, using it on lengthy streams (eg, music) will use a lot of memory. - * Such streams are better handled using getAudioData to keep smaller chunks in - * memory at any given time. */ -void *decodeAudioStream(FilePtr, size_t *length); - -#ifdef __cplusplus -} -#endif /* __cplusplus */ - -#endif /* EXAMPLES_SDL_SOUND_H */ |