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Update OpenAL-soft to 1.23.1-bc7cb17.

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This commit is contained in:
Miku AuahDark
2024-03-20 11:06:03 +08:00
parent 4a512be715
commit 73a6fc9196
294 changed files with 44342 additions and 40077 deletions
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libs/openal-soft/examples/alplay.c
+172 -15
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@@ -37,19 +37,31 @@
#include "common/alhelpers.h"
#include "win_main_utf8.h"
enum FormatType {
Int16,
Float,
IMA4,
MSADPCM
};
/* LoadBuffer loads the named audio file into an OpenAL buffer object, and
* returns the new buffer ID.
*/
static ALuint LoadSound(const char *filename)
{
enum FormatType sample_format = Int16;
ALint byteblockalign = 0;
ALint splblockalign = 0;
sf_count_t num_frames;
ALenum err, format;
ALuint buffer;
ALsizei num_bytes;
SNDFILE *sndfile;
SF_INFO sfinfo;
short *membuf;
sf_count_t num_frames;
ALsizei num_bytes;
ALuint buffer;
void *membuf;
/* Open the audio file and check that it's usable. */
sndfile = sf_open(filename, SFM_READ, &sfinfo);
@@ -58,28 +70,151 @@ static ALuint LoadSound(const char *filename)
fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
return 0;
}
if(sfinfo.frames < 1 || sfinfo.frames > (sf_count_t)(INT_MAX/sizeof(short))/sfinfo.channels)
if(sfinfo.frames < 1)
{
fprintf(stderr, "Bad sample count in %s (%" PRId64 ")\n", filename, sfinfo.frames);
sf_close(sndfile);
return 0;
}
/* Get the sound format, and figure out the OpenAL format */
/* Detect a suitable format to load. Formats like Vorbis and Opus use float
* natively, so load as float to avoid clipping when possible. Formats
* larger than 16-bit can also use float to preserve a bit more precision.
*/
switch((sfinfo.format&SF_FORMAT_SUBMASK))
{
case SF_FORMAT_PCM_24:
case SF_FORMAT_PCM_32:
case SF_FORMAT_FLOAT:
case SF_FORMAT_DOUBLE:
case SF_FORMAT_VORBIS:
case SF_FORMAT_OPUS:
case SF_FORMAT_ALAC_20:
case SF_FORMAT_ALAC_24:
case SF_FORMAT_ALAC_32:
case 0x0080/*SF_FORMAT_MPEG_LAYER_I*/:
case 0x0081/*SF_FORMAT_MPEG_LAYER_II*/:
case 0x0082/*SF_FORMAT_MPEG_LAYER_III*/:
if(alIsExtensionPresent("AL_EXT_FLOAT32"))
sample_format = Float;
break;
case SF_FORMAT_IMA_ADPCM:
/* ADPCM formats require setting a block alignment as specified in the
* file, which needs to be read from the wave 'fmt ' chunk manually
* since libsndfile doesn't provide it in a format-agnostic way.
*/
if(sfinfo.channels <= 2 && (sfinfo.format&SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV
&& alIsExtensionPresent("AL_EXT_IMA4")
&& alIsExtensionPresent("AL_SOFT_block_alignment"))
sample_format = IMA4;
break;
case SF_FORMAT_MS_ADPCM:
if(sfinfo.channels <= 2 && (sfinfo.format&SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV
&& alIsExtensionPresent("AL_SOFT_MSADPCM")
&& alIsExtensionPresent("AL_SOFT_block_alignment"))
sample_format = MSADPCM;
break;
}
if(sample_format == IMA4 || sample_format == MSADPCM)
{
/* For ADPCM, lookup the wave file's "fmt " chunk, which is a
* WAVEFORMATEX-based structure for the audio format.
*/
SF_CHUNK_INFO inf = { "fmt ", 4, 0, NULL };
SF_CHUNK_ITERATOR *iter = sf_get_chunk_iterator(sndfile, &inf);
/* If there's an issue getting the chunk or block alignment, load as
* 16-bit and have libsndfile do the conversion.
*/
if(!iter || sf_get_chunk_size(iter, &inf) != SF_ERR_NO_ERROR || inf.datalen < 14)
sample_format = Int16;
else
{
ALubyte *fmtbuf = calloc(inf.datalen, 1);
inf.data = fmtbuf;
if(sf_get_chunk_data(iter, &inf) != SF_ERR_NO_ERROR)
sample_format = Int16;
else
{
/* Read the nBlockAlign field, and convert from bytes- to
* samples-per-block (verifying it's valid by converting back
* and comparing to the original value).
*/
byteblockalign = fmtbuf[12] | (fmtbuf[13]<<8);
if(sample_format == IMA4)
{
splblockalign = (byteblockalign/sfinfo.channels - 4)/4*8 + 1;
if(splblockalign < 1
|| ((splblockalign-1)/2 + 4)*sfinfo.channels != byteblockalign)
sample_format = Int16;
}
else
{
splblockalign = (byteblockalign/sfinfo.channels - 7)*2 + 2;
if(splblockalign < 2
|| ((splblockalign-2)/2 + 7)*sfinfo.channels != byteblockalign)
sample_format = Int16;
}
}
free(fmtbuf);
}
}
if(sample_format == Int16)
{
splblockalign = 1;
byteblockalign = sfinfo.channels * 2;
}
else if(sample_format == Float)
{
splblockalign = 1;
byteblockalign = sfinfo.channels * 4;
}
/* Figure out the OpenAL format from the file and desired sample type. */
format = AL_NONE;
if(sfinfo.channels == 1)
format = AL_FORMAT_MONO16;
{
if(sample_format == Int16)
format = AL_FORMAT_MONO16;
else if(sample_format == Float)
format = AL_FORMAT_MONO_FLOAT32;
else if(sample_format == IMA4)
format = AL_FORMAT_MONO_IMA4;
else if(sample_format == MSADPCM)
format = AL_FORMAT_MONO_MSADPCM_SOFT;
}
else if(sfinfo.channels == 2)
format = AL_FORMAT_STEREO16;
{
if(sample_format == Int16)
format = AL_FORMAT_STEREO16;
else if(sample_format == Float)
format = AL_FORMAT_STEREO_FLOAT32;
else if(sample_format == IMA4)
format = AL_FORMAT_STEREO_IMA4;
else if(sample_format == MSADPCM)
format = AL_FORMAT_STEREO_MSADPCM_SOFT;
}
else if(sfinfo.channels == 3)
{
if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
format = AL_FORMAT_BFORMAT2D_16;
{
if(sample_format == Int16)
format = AL_FORMAT_BFORMAT2D_16;
else if(sample_format == Float)
format = AL_FORMAT_BFORMAT2D_FLOAT32;
}
}
else if(sfinfo.channels == 4)
{
if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
format = AL_FORMAT_BFORMAT3D_16;
{
if(sample_format == Int16)
format = AL_FORMAT_BFORMAT3D_16;
else if(sample_format == Float)
format = AL_FORMAT_BFORMAT3D_FLOAT32;
}
}
if(!format)
{
@@ -88,10 +223,27 @@ static ALuint LoadSound(const char *filename)
return 0;
}
/* Decode the whole audio file to a buffer. */
membuf = malloc((size_t)(sfinfo.frames * sfinfo.channels) * sizeof(short));
if(sfinfo.frames/splblockalign > (sf_count_t)(INT_MAX/byteblockalign))
{
fprintf(stderr, "Too many samples in %s (%" PRId64 ")\n", filename, sfinfo.frames);
sf_close(sndfile);
return 0;
}
num_frames = sf_readf_short(sndfile, membuf, sfinfo.frames);
/* Decode the whole audio file to a buffer. */
membuf = malloc((size_t)(sfinfo.frames / splblockalign * byteblockalign));
if(sample_format == Int16)
num_frames = sf_readf_short(sndfile, membuf, sfinfo.frames);
else if(sample_format == Float)
num_frames = sf_readf_float(sndfile, membuf, sfinfo.frames);
else
{
sf_count_t count = sfinfo.frames / splblockalign * byteblockalign;
num_frames = sf_read_raw(sndfile, membuf, count);
if(num_frames > 0)
num_frames = num_frames / byteblockalign * splblockalign;
}
if(num_frames < 1)
{
free(membuf);
@@ -99,19 +251,24 @@ static ALuint LoadSound(const char *filename)
fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
return 0;
}
num_bytes = (ALsizei)(num_frames * sfinfo.channels) * (ALsizei)sizeof(short);
num_bytes = (ALsizei)(num_frames / splblockalign * byteblockalign);
printf("Loading: %s (%s, %dhz)\n", filename, FormatName(format), sfinfo.samplerate);
fflush(stdout);
/* Buffer the audio data into a new buffer object, then free the data and
* close the file.
*/
buffer = 0;
alGenBuffers(1, &buffer);
if(splblockalign > 1)
alBufferi(buffer, AL_UNPACK_BLOCK_ALIGNMENT_SOFT, splblockalign);
alBufferData(buffer, format, membuf, num_bytes, sfinfo.samplerate);
free(membuf);
sf_close(sndfile);
/* Check if an error occured, and clean up if so. */
/* Check if an error occurred, and clean up if so. */
err = alGetError();
if(err != AL_NO_ERROR)
{