1 files changed, 210 insertions, 19 deletions

diff --git a/examples/alstream.c b/examples/alstream.c
index 2e427a32..b561406c 100644
--- a/examples/alstream.c
+++ b/examples/alstream.c
@@ -39,20 +39,31 @@
 
 
 /* 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. */
+ * buffers at 200ms each gives a nice per-chunk size, and lets the queue last
+ * for almost one second.
+ */
 #define NUM_BUFFERS 4
-#define BUFFER_SAMPLES 8192
+#define BUFFER_MILLISEC 200
+
+typedef enum SampleType {
+    Int16, Float, IMA4, MSADPCM
+} SampleType;
 
 typedef struct StreamPlayer {
-    /* These are the buffers and source to play out through OpenAL with */
+    /* These are the buffers and source to play out through OpenAL with. */
     ALuint buffers[NUM_BUFFERS];
     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 +78,8 @@ 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;
@@ -112,7 +124,7 @@ 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 +136,148 @@ 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_MPEG_LAYER_I:
+    case SF_FORMAT_MPEG_LAYER_II:
+    case 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)
+            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 +287,9 @@ 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 * BUFFER_MILLISEC / 1000;
+    player->membuf = malloc((size_t)(player->block_count * player->byteblockalign));
 
     return 1;
 }
@@ -162,6 +303,15 @@ static void ClosePlayerFile(StreamPlayer *player)
 
     free(player->membuf);
     player->membuf = NULL;
+
+    if(player->sampleblockalign > 1)
+    {
+        ALsizei i;
+        for(i = 0;i < NUM_BUFFERS;i++)
+            alBufferi(player->buffers[i], AL_UNPACK_BLOCK_ALIGNMENT_SOFT, 0);
+        player->sampleblockalign = 0;
+        player->byteblockalign = 0;
+    }
 }
 
 
@@ -177,11 +327,35 @@ static int StartPlayer(StreamPlayer *player)
     /* Fill the buffer queue */
     for(i = 0;i < NUM_BUFFERS;i++)
     {
+        sf_count_t slen;
+
         /* 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;
+        if(player->sample_type == Int16)
+        {
+            slen = sf_readf_short(player->sndfile, player->membuf,
+                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,
+                player->block_count * player->sampleblockalign);
+            if(slen < 1) break;
+            slen *= player->byteblockalign;
+        }
+        else
+        {
+            slen = sf_read_raw(player->sndfile, player->membuf,
+                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 +401,27 @@ 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,
+                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,
+                player->block_count * player->sampleblockalign);
+            if(slen > 0) slen *= player->byteblockalign;
+        }
+        else
+        {
+            slen = sf_read_raw(player->sndfile, player->membuf,
+                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);