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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/alstream.c
+225 -30
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@@ -37,22 +37,35 @@
#include "common/alhelpers.h"
#include "win_main_utf8.h"
/* Define the number of buffers and buffer size (in milliseconds) to use. 4
* buffers with 8192 samples each gives a nice per-chunk size, and lets the
* queue last for almost one second at 44.1khz. */
#define NUM_BUFFERS 4
#define BUFFER_SAMPLES 8192
* buffers at 200ms each gives a nice per-chunk size, and lets the queue last
* for almost one second.
*/
enum { NumBuffers = 4 };
enum { BufferMillisec = 200 };
typedef enum SampleType {
Int16, Float, IMA4, MSADPCM
} SampleType;
typedef struct StreamPlayer {
/* These are the buffers and source to play out through OpenAL with */
ALuint buffers[NUM_BUFFERS];
/* These are the buffers and source to play out through OpenAL with. */
ALuint buffers[NumBuffers];
ALuint source;
/* Handle for the audio file */
SNDFILE *sndfile;
SF_INFO sfinfo;
short *membuf;
void *membuf;
/* The sample type and block/frame size being read for the buffer. */
SampleType sample_type;
int byteblockalign;
int sampleblockalign;
int block_count;
/* The format of the output stream (sample rate is in sfinfo) */
ALenum format;
@@ -67,7 +80,8 @@ static int UpdatePlayer(StreamPlayer *player);
/* Creates a new player object, and allocates the needed OpenAL source and
* buffer objects. Error checking is simplified for the purposes of this
* example, and will cause an abort if needed. */
* example, and will cause an abort if needed.
*/
static StreamPlayer *NewPlayer(void)
{
StreamPlayer *player;
@@ -76,7 +90,7 @@ static StreamPlayer *NewPlayer(void)
assert(player != NULL);
/* Generate the buffers and source */
alGenBuffers(NUM_BUFFERS, player->buffers);
alGenBuffers(NumBuffers, player->buffers);
assert(alGetError() == AL_NO_ERROR && "Could not create buffers");
alGenSources(1, &player->source);
@@ -99,11 +113,11 @@ static void DeletePlayer(StreamPlayer *player)
ClosePlayerFile(player);
alDeleteSources(1, &player->source);
alDeleteBuffers(NUM_BUFFERS, player->buffers);
alDeleteBuffers(NumBuffers, player->buffers);
if(alGetError() != AL_NO_ERROR)
fprintf(stderr, "Failed to delete object IDs\n");
memset(player, 0, sizeof(*player));
memset(player, 0, sizeof(*player)); /* NOLINT(clang-analyzer-security.insecureAPI.*) */
free(player);
}
@@ -112,7 +126,7 @@ static void DeletePlayer(StreamPlayer *player)
* it will be closed first. */
static int OpenPlayerFile(StreamPlayer *player, const char *filename)
{
size_t frame_size;
int byteblockalign=0, splblockalign=0;
ClosePlayerFile(player);
@@ -124,20 +138,151 @@ static int OpenPlayerFile(StreamPlayer *player, const char *filename)
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((player->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"))
player->sample_type = 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(player->sfinfo.channels <= 2
&& (player->sfinfo.format&SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV
&& alIsExtensionPresent("AL_EXT_IMA4")
&& alIsExtensionPresent("AL_SOFT_block_alignment"))
player->sample_type = IMA4;
break;
case SF_FORMAT_MS_ADPCM:
if(player->sfinfo.channels <= 2
&& (player->sfinfo.format&SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV
&& alIsExtensionPresent("AL_SOFT_MSADPCM")
&& alIsExtensionPresent("AL_SOFT_block_alignment"))
player->sample_type = MSADPCM;
break;
}
if(player->sample_type == IMA4 || player->sample_type == 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(player->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)
player->sample_type = Int16;
else
{
ALubyte *fmtbuf = calloc(inf.datalen, 1);
inf.data = fmtbuf;
if(sf_get_chunk_data(iter, &inf) != SF_ERR_NO_ERROR)
player->sample_type = 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(player->sample_type == IMA4)
{
splblockalign = (byteblockalign/player->sfinfo.channels - 4)/4*8 + 1;
if(splblockalign < 1
|| ((splblockalign-1)/2 + 4)*player->sfinfo.channels != byteblockalign)
player->sample_type = Int16;
}
else
{
splblockalign = (byteblockalign/player->sfinfo.channels - 7)*2 + 2;
if(splblockalign < 2
|| ((splblockalign-2)/2 + 7)*player->sfinfo.channels != byteblockalign)
player->sample_type = Int16;
}
}
free(fmtbuf);
}
}
if(player->sample_type == Int16)
{
player->sampleblockalign = 1;
player->byteblockalign = player->sfinfo.channels * 2;
}
else if(player->sample_type == Float)
{
player->sampleblockalign = 1;
player->byteblockalign = player->sfinfo.channels * 4;
}
else
{
player->sampleblockalign = splblockalign;
player->byteblockalign = byteblockalign;
}
/* Figure out the OpenAL format from the file and desired sample type. */
player->format = AL_NONE;
if(player->sfinfo.channels == 1)
player->format = AL_FORMAT_MONO16;
{
if(player->sample_type == Int16)
player->format = AL_FORMAT_MONO16;
else if(player->sample_type == Float)
player->format = AL_FORMAT_MONO_FLOAT32;
else if(player->sample_type == IMA4)
player->format = AL_FORMAT_MONO_IMA4;
else if(player->sample_type == MSADPCM)
player->format = AL_FORMAT_MONO_MSADPCM_SOFT;
}
else if(player->sfinfo.channels == 2)
player->format = AL_FORMAT_STEREO16;
{
if(player->sample_type == Int16)
player->format = AL_FORMAT_STEREO16;
else if(player->sample_type == Float)
player->format = AL_FORMAT_STEREO_FLOAT32;
else if(player->sample_type == IMA4)
player->format = AL_FORMAT_STEREO_IMA4;
else if(player->sample_type == MSADPCM)
player->format = AL_FORMAT_STEREO_MSADPCM_SOFT;
}
else if(player->sfinfo.channels == 3)
{
if(sf_command(player->sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
player->format = AL_FORMAT_BFORMAT2D_16;
{
if(player->sample_type == Int16)
player->format = AL_FORMAT_BFORMAT2D_16;
else if(player->sample_type == Float)
player->format = AL_FORMAT_BFORMAT2D_FLOAT32;
}
}
else if(player->sfinfo.channels == 4)
{
if(sf_command(player->sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
player->format = AL_FORMAT_BFORMAT3D_16;
{
if(player->sample_type == Int16)
player->format = AL_FORMAT_BFORMAT3D_16;
else if(player->sample_type == Float)
player->format = AL_FORMAT_BFORMAT3D_FLOAT32;
}
}
if(!player->format)
{
@@ -147,8 +292,9 @@ static int OpenPlayerFile(StreamPlayer *player, const char *filename)
return 0;
}
frame_size = (size_t)(BUFFER_SAMPLES * player->sfinfo.channels) * sizeof(short);
player->membuf = malloc(frame_size);
player->block_count = player->sfinfo.samplerate / player->sampleblockalign;
player->block_count = player->block_count * BufferMillisec / 1000;
player->membuf = malloc((size_t)player->block_count * (size_t)player->byteblockalign);
return 1;
}
@@ -162,6 +308,15 @@ static void ClosePlayerFile(StreamPlayer *player)
free(player->membuf);
player->membuf = NULL;
if(player->sampleblockalign > 1)
{
ALsizei i;
for(i = 0;i < NumBuffers;i++)
alBufferi(player->buffers[i], AL_UNPACK_BLOCK_ALIGNMENT_SOFT, 0);
player->sampleblockalign = 0;
player->byteblockalign = 0;
}
}
@@ -175,13 +330,37 @@ static int StartPlayer(StreamPlayer *player)
alSourcei(player->source, AL_BUFFER, 0);
/* Fill the buffer queue */
for(i = 0;i < NUM_BUFFERS;i++)
for(i = 0;i < NumBuffers;i++)
{
/* Get some data to give it to the buffer */
sf_count_t slen = sf_readf_short(player->sndfile, player->membuf, BUFFER_SAMPLES);
if(slen < 1) break;
sf_count_t slen;
/* Get some data to give it to the buffer */
if(player->sample_type == Int16)
{
slen = sf_readf_short(player->sndfile, player->membuf,
(sf_count_t)player->block_count * player->sampleblockalign);
if(slen < 1) break;
slen *= player->byteblockalign;
}
else if(player->sample_type == Float)
{
slen = sf_readf_float(player->sndfile, player->membuf,
(sf_count_t)player->block_count * player->sampleblockalign);
if(slen < 1) break;
slen *= player->byteblockalign;
}
else
{
slen = sf_read_raw(player->sndfile, player->membuf,
(sf_count_t)player->block_count * player->byteblockalign);
if(slen > 0) slen -= slen%player->byteblockalign;
if(slen < 1) break;
}
if(player->sampleblockalign > 1)
alBufferi(player->buffers[i], AL_UNPACK_BLOCK_ALIGNMENT_SOFT,
player->sampleblockalign);
slen *= player->sfinfo.channels * (sf_count_t)sizeof(short);
alBufferData(player->buffers[i], player->format, player->membuf, (ALsizei)slen,
player->sfinfo.samplerate);
}
@@ -227,10 +406,27 @@ static int UpdatePlayer(StreamPlayer *player)
/* Read the next chunk of data, refill the buffer, and queue it
* back on the source */
slen = sf_readf_short(player->sndfile, player->membuf, BUFFER_SAMPLES);
if(player->sample_type == Int16)
{
slen = sf_readf_short(player->sndfile, player->membuf,
(sf_count_t)player->block_count * player->sampleblockalign);
if(slen > 0) slen *= player->byteblockalign;
}
else if(player->sample_type == Float)
{
slen = sf_readf_float(player->sndfile, player->membuf,
(sf_count_t)player->block_count * player->sampleblockalign);
if(slen > 0) slen *= player->byteblockalign;
}
else
{
slen = sf_read_raw(player->sndfile, player->membuf,
(sf_count_t)player->block_count * player->byteblockalign);
if(slen > 0) slen -= slen%player->byteblockalign;
}
if(slen > 0)
{
slen *= player->sfinfo.channels * (sf_count_t)sizeof(short);
alBufferData(bufid, player->format, player->membuf, (ALsizei)slen,
player->sfinfo.samplerate);
alSourceQueueBuffers(player->source, 1, &bufid);
@@ -292,10 +488,9 @@ int main(int argc, char **argv)
/* Get the name portion, without the path, for display. */
namepart = strrchr(argv[i], '/');
if(namepart || (namepart=strrchr(argv[i], '\\')))
namepart++;
else
namepart = argv[i];
if(!namepart) namepart = strrchr(argv[i], '\\');
if(!namepart) namepart = argv[i];
else namepart++;
printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->format),
player->sfinfo.samplerate);