1 files changed, 201 insertions, 44 deletions

diff --git a/examples/alplay.c b/examples/alplay.c
index 09ad96b4..4291cb47 100644
--- a/examples/alplay.c
+++ b/examples/alplay.c
@@ -24,85 +24,247 @@
 
 /* This file contains an example for playing a sound buffer. */
 
-#include <stdio.h>
 #include <assert.h>
+#include <inttypes.h>
+#include <limits.h>
+#include <stdio.h>
+#include <stdlib.h>
 
-#include "SDL_sound.h"
-#include "SDL_audio.h"
-#include "SDL_stdinc.h"
+#include "sndfile.h"
 
 #include "AL/al.h"
+#include "AL/alext.h"
 
 #include "common/alhelpers.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)
 {
-    Sound_Sample *sample;
+    enum FormatType sample_format = Int16;
+    ALint byteblockalign = 0;
+    ALint splblockalign = 0;
+    sf_count_t num_frames;
     ALenum err, format;
+    ALsizei num_bytes;
+    SNDFILE *sndfile;
+    SF_INFO sfinfo;
     ALuint buffer;
-    Uint32 slen;
+    void *membuf;
 
-    /* Open the audio file */
-    sample = Sound_NewSampleFromFile(filename, NULL, 65536);
-    if(!sample)
+    /* Open the audio file and check that it's usable. */
+    sndfile = sf_open(filename, SFM_READ, &sfinfo);
+    if(!sndfile)
+    {
+        fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
+        return 0;
+    }
+    if(sfinfo.frames < 1)
     {
-        fprintf(stderr, "Could not open audio in %s\n", filename);
+        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 */
-    if(sample->actual.channels == 1)
+    /* 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))
     {
-        if(sample->actual.format == AUDIO_U8)
-            format = AL_FORMAT_MONO8;
-        else if(sample->actual.format == AUDIO_S16SYS)
-            format = AL_FORMAT_MONO16;
+    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
         {
-            fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
-            Sound_FreeSample(sample);
-            return 0;
+            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);
         }
     }
-    else if(sample->actual.channels == 2)
+
+    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)
+    {
+        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)
     {
-        if(sample->actual.format == AUDIO_U8)
-            format = AL_FORMAT_STEREO8;
-        else if(sample->actual.format == AUDIO_S16SYS)
+        if(sample_format == Int16)
             format = AL_FORMAT_STEREO16;
-        else
+        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)
         {
-            fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
-            Sound_FreeSample(sample);
-            return 0;
+            if(sample_format == Int16)
+                format = AL_FORMAT_BFORMAT2D_16;
+            else if(sample_format == Float)
+                format = AL_FORMAT_BFORMAT2D_FLOAT32;
         }
     }
-    else
+    else if(sfinfo.channels == 4)
+    {
+        if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
+        {
+            if(sample_format == Int16)
+                format = AL_FORMAT_BFORMAT3D_16;
+            else if(sample_format == Float)
+                format = AL_FORMAT_BFORMAT3D_FLOAT32;
+        }
+    }
+    if(!format)
     {
-        fprintf(stderr, "Unsupported channel count: %d\n", sample->actual.channels);
-        Sound_FreeSample(sample);
+        fprintf(stderr, "Unsupported channel count: %d\n", sfinfo.channels);
+        sf_close(sndfile);
         return 0;
     }
 
-    /* Decode the whole audio stream to a buffer. */
-    slen = Sound_DecodeAll(sample);
-    if(!sample->buffer || slen == 0)
+    if(sfinfo.frames/splblockalign > (sf_count_t)(INT_MAX/byteblockalign))
     {
-        fprintf(stderr, "Failed to read audio from %s\n", filename);
-        Sound_FreeSample(sample);
+        fprintf(stderr, "Too many samples in %s (%" PRId64 ")\n", filename, sfinfo.frames);
+        sf_close(sndfile);
         return 0;
     }
 
+    /* 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);
+        sf_close(sndfile);
+        fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
+        return 0;
+    }
+    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. */
+     * close the file.
+     */
     buffer = 0;
     alGenBuffers(1, &buffer);
-    alBufferData(buffer, format, sample->buffer, (ALsizei)slen, (ALsizei)sample->actual.rate);
-    Sound_FreeSample(sample);
+    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. */
     err = alGetError();
@@ -136,14 +298,10 @@ int main(int argc, char **argv)
     if(InitAL(&argv, &argc) != 0)
         return 1;
 
-    /* Initialize SDL_sound. */
-    Sound_Init();
-
     /* Load the sound into a buffer. */
     buffer = LoadSound(argv[0]);
     if(!buffer)
     {
-        Sound_Quit();
         CloseAL();
         return 1;
     }
@@ -167,11 +325,10 @@ int main(int argc, char **argv)
     } while(alGetError() == AL_NO_ERROR && state == AL_PLAYING);
     printf("\n");
 
-    /* All done. Delete resources, and close down SDL_sound and OpenAL. */
+    /* All done. Delete resources, and close down OpenAL. */
     alDeleteSources(1, &source);
     alDeleteBuffers(1, &buffer);
 
-    Sound_Quit();
     CloseAL();
 
     return 0;