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Simplify OpenAL code

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This commit is contained in:
Aaron Culliney 2014-10-12 10:31:10 -07:00
parent 7f395edc11
commit 13d17af838
3 changed files with 5 additions and 259 deletions
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177
src/audio/alhelpers.c
View File

@ -87,183 +87,6 @@ void CloseAL(void)
alcCloseDevice(device); alcCloseDevice(device);
} }
/* 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)
{
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) const char *ChannelsName(ALenum chans)
{ {
switch(chans) switch(chans)

7
src/audio/alhelpers.h
View File

@ -21,13 +21,6 @@ const char *FormatName(ALenum format);
ALsizei FramesToBytes(ALsizei size, ALenum channels, ALenum type); ALsizei FramesToBytes(ALsizei size, ALenum channels, ALenum type);
ALsizei BytesToFrames(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 /* Loads samples into a buffer using the standard alBufferData call, but with a
* LPALBUFFERSAMPLESSOFT-compatible prototype. Assumes internalformat is valid * LPALBUFFERSAMPLESSOFT-compatible prototype. Assumes internalformat is valid
* for alBufferData, and that channels and type match it. */ * for alBufferData, and that channels and type match it. */

80
src/audio/soundcore-openal.c
View File

@ -18,9 +18,6 @@
#include "audio/soundcore-openal.h" #include "audio/soundcore-openal.h"
#include "audio/alhelpers.h" #include "audio/alhelpers.h"
LPALBUFFERSAMPLESSOFT alBufferSamplesSOFT = wrap_BufferSamples;
LPALISBUFFERFORMATSUPPORTEDSOFT alIsBufferFormatSupportedSOFT = NULL;
static long OpenALCreateSoundBuffer(ALBufferParamsStruct *params, ALSoundBufferStruct **soundbuf_struct, void *extra_data); static long OpenALCreateSoundBuffer(ALBufferParamsStruct *params, ALSoundBufferStruct **soundbuf_struct, void *extra_data);
static long OpenALDestroySoundBuffer(ALSoundBufferStruct **soundbuf_struct); static long OpenALDestroySoundBuffer(ALSoundBufferStruct **soundbuf_struct);
@ -128,8 +125,6 @@ long SoundSystemCreate(const char *sound_device, SoundSystemStruct **sound_struc
if (alIsExtensionPresent("AL_SOFT_buffer_samples")) if (alIsExtensionPresent("AL_SOFT_buffer_samples"))
{ {
LOG("AL_SOFT_buffer_samples supported, good!"); LOG("AL_SOFT_buffer_samples supported, good!");
alBufferSamplesSOFT = alGetProcAddress("alBufferSamplesSOFT");
alIsBufferFormatSupportedSOFT = alGetProcAddress("alIsBufferFormatSupportedSOFT");
} }
else else
{ {
@ -340,22 +335,14 @@ static ALVoice *NewVoice(ALBufferParamsStruct *params)
// Emulator supports only mono and stereo // Emulator supports only mono and stereo
if (params->lpwfxFormat->nChannels == 2) if (params->lpwfxFormat->nChannels == 2)
{ {
voice->channels = AL_STEREO_SOFT; voice->format = AL_FORMAT_STEREO16;
} }
else else
{ {
voice->channels = AL_MONO_SOFT; voice->format = AL_FORMAT_MONO16;
}
voice->type = AL_SHORT_SOFT; // signed 16bit
voice->format = GetFormat(voice->channels, voice->type, alIsBufferFormatSupportedSOFT);
if (voice->format == 0)
{
ERRLOG("OOPS, Unsupported format (%s, %s)", ChannelsName(voice->channels), TypeName(voice->type));
break;
} }
/* Allocate enough space for the temp buffer, given the format */ /* Allocate enough space for the temp buffer, given the format */
//voice->buffersize = FramesToBytes(params->dwBufferBytes, voice->channels, voice->type);
voice->buffersize = params->dwBufferBytes; voice->buffersize = params->dwBufferBytes;
voice->data = malloc(voice->buffersize); voice->data = malloc(voice->buffersize);
if (voice->data == NULL) if (voice->data == NULL)
@ -364,12 +351,9 @@ static ALVoice *NewVoice(ALBufferParamsStruct *params)
break; break;
} }
LOG("\tType : 0x%08x", voice->type);
LOG("\tRate : 0x%08x", voice->rate); LOG("\tRate : 0x%08x", voice->rate);
LOG("\tChannels : 0x%08x", voice->channels);
LOG("\tFormat : 0x%08x", voice->format); LOG("\tFormat : 0x%08x", voice->format);
LOG("\tbuffersize : %d", voice->buffersize); LOG("\tbuffersize : %d", voice->buffersize);
LOG("\tSOFT : %s", alIsBufferFormatSupportedSOFT ? "YES" : "NO");
return voice; return voice;
@ -417,59 +401,6 @@ static long ALRestore(void *_this)
static long ALPlay(void *_this, unsigned long reserved1, unsigned long reserved2, unsigned long flags) static long ALPlay(void *_this, unsigned long reserved1, unsigned long reserved2, unsigned long flags)
{ {
LOG("ALPlay ..."); LOG("ALPlay ...");
#if 0
ALVoice *voice = (ALVoice*)_this;
int err = 0;
// Rewind the source position and clear the buffer queue
alSourceRewind(voice->source);
if((err = alGetError()) != AL_NO_ERROR)
{
ERRLOG("OOPS, alSourceRewind : 0x%08x", err);
return err;
}
alSourcei(voice->source, AL_BUFFER, 0);
if((err = alGetError()) != AL_NO_ERROR)
{
ERRLOG("OOPS, alSourcei : 0x%08x", err);
return err;
}
if (false) {
// fill buffer queue with quiet
memset(voice->data, 0x0, voice->buffersize);
int half_buffers = (OPENAL_NUM_BUFFERS>>1);
for (size_t i=0; i<half_buffers; i++)
{
alBufferSamplesSOFT(voice->buffers[i], voice->rate, voice->format,
BytesToFrames(voice->buffersize, voice->channels, voice->type),
voice->channels, voice->type, voice->data);
err = alGetError();
if (err != AL_NO_ERROR)
{
LOG("Error buffering initial samples : 0x%08x", err);
return err;
}
}
alSourceQueueBuffers(voice->source, OPENAL_NUM_BUFFERS, voice->buffers);
err = alGetError();
if (err != AL_NO_ERROR)
{
LOG("Error queueing initial buffers : 0x%08x", err);
return err;
}
alSourcePlay(voice->source);
err = alGetError();
if (err != AL_NO_ERROR)
{
LOG("Error starting playback : 0x%08x", err);
return err;
}
}
#endif
return 0; return 0;
} }
@ -616,13 +547,12 @@ static long _ALSubmitBufferToOpenAL(ALVoice *voice)
return -1; return -1;
} }
//LOG("Enqueing OpenAL buffer %u (%u bytes)", node->bufid, node->bytes); //LOG("Enqueing OpenAL buffer %u (%u bytes)", node->bufid, node->bytes);
alBufferSamplesSOFT(node->bufid, voice->rate, voice->format, alBufferData(node->bufid, voice->format, voice->data, node->bytes, voice->rate);
BytesToFrames(node->bytes, voice->channels, voice->type),
voice->channels, voice->type, voice->data);
if ((err = alGetError()) != AL_NO_ERROR) if ((err = alGetError()) != AL_NO_ERROR)
{ {
PlaylistDequeue(voice, node); PlaylistDequeue(voice, node);
ERRLOG("OOPS, Error alBufferSamplesSOFT : 0x%08x", err); ERRLOG("OOPS, Error alBufferData : 0x%08x", err);
return err; return err;
} }