audio: add a few enhancements

MAKE.bat

@@ -33,6 +33,7 @@ if "%1"=="help" (
     echo %0 [v4web]                 ; sync v4 website
     echo %0 [swap]                  ; toggle #line directives on/off
     echo %0 [split^|join]            ; engine/v4k* ^>split^> engine/split/* or engine/split/* ^>join^> engine/v4k*
+    echo %0 [3rd]                   ; join 3rd parties together
     echo %0 [lua]                   ; execute lua script with v4k
     echo %0 [amalgamation]          ; combine engine/v4k* into a single-header file
     echo %0 [prep]                  ; combine split files into a single-header file, ready for use
@@ -253,6 +254,10 @@ if "%1"=="join" (
     call tools\join
     exit /b
 )
+if "%1"=="3rd" (
+    call tools\join_3rd
+    exit /b
+)
 if "%1"=="swap" (
     echo Swapping #line on v4k.h
     call tools\linswap engine\v4k.h

bind/v4k.lua

@@ -2083,6 +2083,8 @@ typedef struct audio_handle* audio_t;
  int     audio_play_gain_pitch( audio_t a, int flags, float gain, float pitch );
  int     audio_play_gain_pitch_pan( audio_t a, int flags, float gain, float pitch, float pan );
  int     audio_stop( audio_t a );
+ void    audio_loop( audio_t a, bool loop );
+ bool    audio_playing( audio_t a );
  float   audio_volume_clip(float gain);
  float   audio_volume_stream(float gain);
  float   audio_volume_master(float gain);
@@ -3349,7 +3351,7 @@ unsigned play;
 bool paused;
 struct atlas_t *a;
 } sprite_t;
-enum { OBJTYPE_sprite_t = 10 };       typedef struct { unsigned static_assert_on_L__LINE__ : !!(10 <= 255); } static_assert_on_Lconcat(_L,3973)___COUNTER__;       typedef struct { unsigned static_assert_on_L__LINE__ : !!(sizeof(sprite_t)); } static_assert_on_Lconcat(_L,3973)___COUNTER__;;
+enum { OBJTYPE_sprite_t = 10 };       typedef struct { unsigned static_assert_on_L__LINE__ : !!(10 <= 255); } static_assert_on_Lconcat(_L,3975)___COUNTER__;       typedef struct { unsigned static_assert_on_L__LINE__ : !!(sizeof(sprite_t)); } static_assert_on_Lconcat(_L,3975)___COUNTER__;;
  void     sprite_ctor(sprite_t *s);
  void     sprite_dtor(sprite_t *s);
  void     sprite_tick(sprite_t *s);

engine/joint/v4k.h

@@ -15636,6 +15636,8 @@ API int     audio_play_gain( audio_t a, int flags, float gain/*0*/ );
 API int     audio_play_gain_pitch( audio_t a, int flags, float gain, float pitch/*1*/ );
 API int     audio_play_gain_pitch_pan( audio_t a, int flags, float gain, float pitch, float pan/*0*/ );
 API int     audio_stop( audio_t a );
+API void    audio_loop( audio_t a, bool loop );
+API bool    audio_playing( audio_t a );
 
 API float   audio_volume_clip(float gain);   // set     fx volume if gain is in [0..1] range. returns current     fx volume in any case
 API float   audio_volume_stream(float gain); // set    bgm volume if gain is in [0..1] range. returns current    bgm volume in any case
@@ -78392,6 +78394,7 @@ WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 //    See the example at the end of the file.
 //
 //  VERSION HISTORY
+//    0.03 (2022-12-12) add an ability to stop audio stream via callback, add a method to check if voice is already stopped
 //    0.02 (2022-05-10) allow voice queueing in same channel. ie, chain another sample on same voice channel after current sample playback is done (@r-lyeh)
 //    0.01 (2016-05-01) initial version
 //
@@ -78440,7 +78443,7 @@ typedef struct {
 //
 
 // The callback which will be called when the stream needs more data.
-typedef void (*sts_mixer_stream_callback)(sts_mixer_sample_t* sample, void* userdata);
+typedef bool (*sts_mixer_stream_callback)(sts_mixer_sample_t* sample, void* userdata);
 
 typedef struct {
   void*                     userdata;         // a userdata pointer which will passed to the callback
@@ -78508,6 +78511,12 @@ int sts_mixer_play_stream(sts_mixer_t* mixer, sts_mixer_stream_t* stream, float
 // Stops voice with the given voice no. You can pass the returned number of sts_mixer_play_sample / sts_mixer_play_stream here.
 void sts_mixer_stop_voice(sts_mixer_t* mixer, int voice);
 
+// Returns whether the given sample has already stopped playing.
+bool sts_mixer_sample_stopped(sts_mixer_t* mixer, sts_mixer_sample_t* sample);
+
+// Returns whether the given stream has already stopped playing.
+bool sts_mixer_stream_stopped(sts_mixer_t* mixer, sts_mixer_stream_t* stream);
+
 // Stops all voices playing the given sample. Useful when you want to delete the sample and make sure it is not used anymore.
 void sts_mixer_stop_sample(sts_mixer_t* mixer, sts_mixer_sample_t* sample);
 
@@ -78650,6 +78659,19 @@ void sts_mixer_stop_voice(sts_mixer_t* mixer, int voice) {
   if (voice >= 0 && voice < STS_MIXER_VOICES) sts_mixer__reset_voice(mixer, voice);
 }
 
+bool sts_mixer_sample_stopped(sts_mixer_t* mixer, sts_mixer_sample_t* sample) {
+  for (int i = 0; i < STS_MIXER_VOICES; ++i) {
+    if (mixer->voices[i].sample == sample && mixer->voices[i].state != STS_MIXER_VOICE_STOPPED) return false;
+  }
+  return true;
+}
+
+bool sts_mixer_stream_stopped(sts_mixer_t* mixer, sts_mixer_stream_t* stream) {
+  for (int i = 0; i < STS_MIXER_VOICES; ++i) {
+    if (mixer->voices[i].stream == stream && mixer->voices[i].state != STS_MIXER_VOICE_STOPPED) return false;
+  }
+  return true;
+}
 
 void sts_mixer_stop_sample(sts_mixer_t* mixer, sts_mixer_sample_t* sample) {
   int i;
@@ -78699,9 +78721,13 @@ void sts_mixer_mix_audio(sts_mixer_t* mixer, void* output, unsigned int samples)
         position = ((int)voice->position) * 2;
         if (position >= voice->stream->sample.length) {
           // buffer empty...refill
-          voice->stream->callback(&voice->stream->sample, voice->stream->userdata);
+          if (voice->stream->callback(&voice->stream->sample, voice->stream->userdata)) {
             voice->position = 0.0f;
             position = 0;
+          } else {
+            sts_mixer__reset_voice(mixer, i);
+            continue;
+          }
         }
         left += sts_mixer__clamp_sample(sts_mixer__get_sample(&voice->stream->sample, position) * voice->gain);
         right += sts_mixer__clamp_sample(sts_mixer__get_sample(&voice->stream->sample, position + 1) * voice->gain);
@@ -353847,6 +353873,7 @@ typedef struct {
     float               dataf[4096*2];
     };
     bool rewind;
+    bool loop;
 } mystream_t;
 
 static void downsample_to_mono_flt( int channels, float *buffer, int samples ) {
@@ -353871,7 +353898,7 @@ static void downsample_to_mono_s16( int channels, short *buffer, int samples ) {
 }
 
 // the callback to refill the (stereo) stream data
-static void refill_stream(sts_mixer_sample_t* sample, void* userdata) {
+static bool refill_stream(sts_mixer_sample_t* sample, void* userdata) {
     mystream_t* stream = (mystream_t*)userdata;
     switch( stream->type ) {
         default:
@@ -353880,6 +353907,7 @@ static void refill_stream(sts_mixer_sample_t* sample, void* userdata) {
             if( stream->rewind ) stream->rewind = 0, ma_dr_wav_seek_to_pcm_frame(&stream->wav, 0);
             if (ma_dr_wav_read_pcm_frames_s16(&stream->wav, sl, (short*)stream->data) < sl) {
                 ma_dr_wav_seek_to_pcm_frame(&stream->wav, 0);
+                if (!stream->loop) return false;
             }
         }
         break; case MP3: {
@@ -353887,6 +353915,7 @@ static void refill_stream(sts_mixer_sample_t* sample, void* userdata) {
             if( stream->rewind ) stream->rewind = 0, ma_dr_mp3_seek_to_pcm_frame(&stream->mp3_, 0);
             if (ma_dr_mp3_read_pcm_frames_f32(&stream->mp3_, sl, stream->dataf) < sl) {
                 ma_dr_mp3_seek_to_pcm_frame(&stream->mp3_, 0);
+                if (!stream->loop) return false;
             }
         }
         break; case OGG: {
@@ -353894,9 +353923,12 @@ static void refill_stream(sts_mixer_sample_t* sample, void* userdata) {
             if( stream->rewind ) stream->rewind = 0, stb_vorbis_seek(stream->ogg, 0);
             if( stb_vorbis_get_samples_short_interleaved(ogg, 2, (short*)stream->data, sample->length) == 0 )  {
                 stb_vorbis_seek(stream->ogg, 0);
+                if (!stream->loop) return false;
             }
         }
     }
+
+    return true;
 }
 static void reset_stream(mystream_t* stream) {
     if( stream ) memset( stream->data, 0, sizeof(stream->data) ), stream->rewind = 1;
@@ -353910,6 +353942,7 @@ static bool load_stream(mystream_t* stream, const char *filename) {
     int error;
     int HZ = 44100;
     stream->type = UNK;
+    stream->loop = true;
     if( stream->type == UNK && (stream->ogg = stb_vorbis_open_memory((const unsigned char *)data, datalen, &error, NULL)) ) {
         stb_vorbis_info info = stb_vorbis_get_info(stream->ogg);
         if( info.channels != 2 ) { puts("cannot stream ogg file. stereo required."); goto end; } // @fixme: upsample
@@ -354221,6 +354254,22 @@ int audio_stop( audio_t a ) {
     return 1;
 }
 
+void audio_loop( audio_t a, bool loop ) {
+    if ( a->is_stream ) {
+        a->stream.loop = loop;
+    }
+}
+
+bool audio_playing( audio_t a ) {
+    if( a->is_clip ) {
+        return !sts_mixer_sample_stopped(&mixer, &a->clip);
+    }
+    if( a->is_stream ) {
+        return !sts_mixer_stream_stopped(&mixer, &a->stream.stream);
+    }
+    return false;
+}
+
 // -----------------------------------------------------------------------------
 // audio queue
 
@@ -354248,7 +354297,7 @@ static void audio_queue_init() {
     do_once thread_queue_init(&queue_mutex, countof(audio_queues), audio_queues, 0);
 }
 
-static void audio_queue_callback(sts_mixer_sample_t* sample, void* userdata) {
+static bool audio_queue_callback(sts_mixer_sample_t* sample, void* userdata) {
     (void)userdata;
 
     int sl = sample->length / 2; // 2 ch
@@ -354272,6 +354321,8 @@ static void audio_queue_callback(sts_mixer_sample_t* sample, void* userdata) {
             aq = 0;
         }
     } while( bytes > 0 );
+
+    return 1;
 }
 
 static int audio_queue_voice = -1;

engine/split/3rd_sts_mixer.h

@@ -18,6 +18,7 @@
 //    See the example at the end of the file.
 //
 //  VERSION HISTORY
+//    0.03 (2022-12-12) add an ability to stop audio stream via callback, add a method to check if voice is already stopped
 //    0.02 (2022-05-10) allow voice queueing in same channel. ie, chain another sample on same voice channel after current sample playback is done (@r-lyeh)
 //    0.01 (2016-05-01) initial version
 //
@@ -66,7 +67,7 @@ typedef struct {
 //
 
 // The callback which will be called when the stream needs more data.
-typedef void (*sts_mixer_stream_callback)(sts_mixer_sample_t* sample, void* userdata);
+typedef bool (*sts_mixer_stream_callback)(sts_mixer_sample_t* sample, void* userdata);
 
 typedef struct {
   void*                     userdata;         // a userdata pointer which will passed to the callback
@@ -134,6 +135,12 @@ int sts_mixer_play_stream(sts_mixer_t* mixer, sts_mixer_stream_t* stream, float
 // Stops voice with the given voice no. You can pass the returned number of sts_mixer_play_sample / sts_mixer_play_stream here.
 void sts_mixer_stop_voice(sts_mixer_t* mixer, int voice);
 
+// Returns whether the given sample has already stopped playing.
+bool sts_mixer_sample_stopped(sts_mixer_t* mixer, sts_mixer_sample_t* sample);
+
+// Returns whether the given stream has already stopped playing.
+bool sts_mixer_stream_stopped(sts_mixer_t* mixer, sts_mixer_stream_t* stream);
+
 // Stops all voices playing the given sample. Useful when you want to delete the sample and make sure it is not used anymore.
 void sts_mixer_stop_sample(sts_mixer_t* mixer, sts_mixer_sample_t* sample);
 
@@ -276,6 +283,19 @@ void sts_mixer_stop_voice(sts_mixer_t* mixer, int voice) {
   if (voice >= 0 && voice < STS_MIXER_VOICES) sts_mixer__reset_voice(mixer, voice);
 }
 
+bool sts_mixer_sample_stopped(sts_mixer_t* mixer, sts_mixer_sample_t* sample) {
+  for (int i = 0; i < STS_MIXER_VOICES; ++i) {
+    if (mixer->voices[i].sample == sample && mixer->voices[i].state != STS_MIXER_VOICE_STOPPED) return false;
+  }
+  return true;
+}
+
+bool sts_mixer_stream_stopped(sts_mixer_t* mixer, sts_mixer_stream_t* stream) {
+  for (int i = 0; i < STS_MIXER_VOICES; ++i) {
+    if (mixer->voices[i].stream == stream && mixer->voices[i].state != STS_MIXER_VOICE_STOPPED) return false;
+  }
+  return true;
+}
 
 void sts_mixer_stop_sample(sts_mixer_t* mixer, sts_mixer_sample_t* sample) {
   int i;
@@ -325,9 +345,13 @@ void sts_mixer_mix_audio(sts_mixer_t* mixer, void* output, unsigned int samples)
         position = ((int)voice->position) * 2;
         if (position >= voice->stream->sample.length) {
           // buffer empty...refill
-          voice->stream->callback(&voice->stream->sample, voice->stream->userdata);
+          if (voice->stream->callback(&voice->stream->sample, voice->stream->userdata)) {
             voice->position = 0.0f;
             position = 0;
+          } else {
+            sts_mixer__reset_voice(mixer, i);
+            continue;
+          }
         }
         left += sts_mixer__clamp_sample(sts_mixer__get_sample(&voice->stream->sample, position) * voice->gain);
         right += sts_mixer__clamp_sample(sts_mixer__get_sample(&voice->stream->sample, position + 1) * voice->gain);

engine/split/v4k_audio.c

@@ -70,6 +70,7 @@ typedef struct {
     float               dataf[4096*2];
     };
     bool rewind;
+    bool loop;
 } mystream_t;
 
 static void downsample_to_mono_flt( int channels, float *buffer, int samples ) {
@@ -94,7 +95,7 @@ static void downsample_to_mono_s16( int channels, short *buffer, int samples ) {
 }
 
 // the callback to refill the (stereo) stream data
-static void refill_stream(sts_mixer_sample_t* sample, void* userdata) {
+static bool refill_stream(sts_mixer_sample_t* sample, void* userdata) {
     mystream_t* stream = (mystream_t*)userdata;
     switch( stream->type ) {
         default:
@@ -103,6 +104,7 @@ static void refill_stream(sts_mixer_sample_t* sample, void* userdata) {
             if( stream->rewind ) stream->rewind = 0, ma_dr_wav_seek_to_pcm_frame(&stream->wav, 0);
             if (ma_dr_wav_read_pcm_frames_s16(&stream->wav, sl, (short*)stream->data) < sl) {
                 ma_dr_wav_seek_to_pcm_frame(&stream->wav, 0);
+                if (!stream->loop) return false;
             }
         }
         break; case MP3: {
@@ -110,6 +112,7 @@ static void refill_stream(sts_mixer_sample_t* sample, void* userdata) {
             if( stream->rewind ) stream->rewind = 0, ma_dr_mp3_seek_to_pcm_frame(&stream->mp3_, 0);
             if (ma_dr_mp3_read_pcm_frames_f32(&stream->mp3_, sl, stream->dataf) < sl) {
                 ma_dr_mp3_seek_to_pcm_frame(&stream->mp3_, 0);
+                if (!stream->loop) return false;
             }
         }
         break; case OGG: {
@@ -117,9 +120,12 @@ static void refill_stream(sts_mixer_sample_t* sample, void* userdata) {
             if( stream->rewind ) stream->rewind = 0, stb_vorbis_seek(stream->ogg, 0);
             if( stb_vorbis_get_samples_short_interleaved(ogg, 2, (short*)stream->data, sample->length) == 0 )  {
                 stb_vorbis_seek(stream->ogg, 0);
+                if (!stream->loop) return false;
             }
         }
     }
+
+    return true;
 }
 static void reset_stream(mystream_t* stream) {
     if( stream ) memset( stream->data, 0, sizeof(stream->data) ), stream->rewind = 1;
@@ -133,6 +139,7 @@ static bool load_stream(mystream_t* stream, const char *filename) {
     int error;
     int HZ = 44100;
     stream->type = UNK;
+    stream->loop = true;
     if( stream->type == UNK && (stream->ogg = stb_vorbis_open_memory((const unsigned char *)data, datalen, &error, NULL)) ) {
         stb_vorbis_info info = stb_vorbis_get_info(stream->ogg);
         if( info.channels != 2 ) { puts("cannot stream ogg file. stereo required."); goto end; } // @fixme: upsample
@@ -444,6 +451,22 @@ int audio_stop( audio_t a ) {
     return 1;
 }
 
+void audio_loop( audio_t a, bool loop ) {
+    if ( a->is_stream ) {
+        a->stream.loop = loop;
+    }
+}
+
+bool audio_playing( audio_t a ) {
+    if( a->is_clip ) {
+        return !sts_mixer_sample_stopped(&mixer, &a->clip);
+    }
+    if( a->is_stream ) {
+        return !sts_mixer_stream_stopped(&mixer, &a->stream.stream);
+    }
+    return false;
+}
+
 // -----------------------------------------------------------------------------
 // audio queue
 
@@ -471,7 +494,7 @@ static void audio_queue_init() {
     do_once thread_queue_init(&queue_mutex, countof(audio_queues), audio_queues, 0);
 }
 
-static void audio_queue_callback(sts_mixer_sample_t* sample, void* userdata) {
+static bool audio_queue_callback(sts_mixer_sample_t* sample, void* userdata) {
     (void)userdata;
 
     int sl = sample->length / 2; // 2 ch
@@ -495,6 +518,8 @@ static void audio_queue_callback(sts_mixer_sample_t* sample, void* userdata) {
             aq = 0;
         }
     } while( bytes > 0 );
+
+    return 1;
 }
 
 static int audio_queue_voice = -1;

engine/split/v4k_audio.h

@@ -23,6 +23,8 @@ API int     audio_play_gain( audio_t a, int flags, float gain/*0*/ );
 API int     audio_play_gain_pitch( audio_t a, int flags, float gain, float pitch/*1*/ );
 API int     audio_play_gain_pitch_pan( audio_t a, int flags, float gain, float pitch, float pan/*0*/ );
 API int     audio_stop( audio_t a );
+API void    audio_loop( audio_t a, bool loop );
+API bool    audio_playing( audio_t a );
 
 API float   audio_volume_clip(float gain);   // set     fx volume if gain is in [0..1] range. returns current     fx volume in any case
 API float   audio_volume_stream(float gain); // set    bgm volume if gain is in [0..1] range. returns current    bgm volume in any case

engine/v4k

@@ -59542,6 +59542,7 @@ WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 //    See the example at the end of the file.
 //
 //  VERSION HISTORY
+//    0.03 (2022-12-12) add an ability to stop audio stream via callback, add a method to check if voice is already stopped
 //    0.02 (2022-05-10) allow voice queueing in same channel. ie, chain another sample on same voice channel after current sample playback is done (@r-lyeh)
 //    0.01 (2016-05-01) initial version
 //
@@ -59590,7 +59591,7 @@ typedef struct {
 //
 
 // The callback which will be called when the stream needs more data.
-typedef void (*sts_mixer_stream_callback)(sts_mixer_sample_t* sample, void* userdata);
+typedef bool (*sts_mixer_stream_callback)(sts_mixer_sample_t* sample, void* userdata);
 
 typedef struct {
   void*                     userdata;         // a userdata pointer which will passed to the callback
@@ -59658,6 +59659,12 @@ int sts_mixer_play_stream(sts_mixer_t* mixer, sts_mixer_stream_t* stream, float
 // Stops voice with the given voice no. You can pass the returned number of sts_mixer_play_sample / sts_mixer_play_stream here.
 void sts_mixer_stop_voice(sts_mixer_t* mixer, int voice);
 
+// Returns whether the given sample has already stopped playing.
+bool sts_mixer_sample_stopped(sts_mixer_t* mixer, sts_mixer_sample_t* sample);
+
+// Returns whether the given stream has already stopped playing.
+bool sts_mixer_stream_stopped(sts_mixer_t* mixer, sts_mixer_stream_t* stream);
+
 // Stops all voices playing the given sample. Useful when you want to delete the sample and make sure it is not used anymore.
 void sts_mixer_stop_sample(sts_mixer_t* mixer, sts_mixer_sample_t* sample);
 
@@ -59800,6 +59807,19 @@ void sts_mixer_stop_voice(sts_mixer_t* mixer, int voice) {
   if (voice >= 0 && voice < STS_MIXER_VOICES) sts_mixer__reset_voice(mixer, voice);
 }
 
+bool sts_mixer_sample_stopped(sts_mixer_t* mixer, sts_mixer_sample_t* sample) {
+  for (int i = 0; i < STS_MIXER_VOICES; ++i) {
+    if (mixer->voices[i].sample == sample && mixer->voices[i].state != STS_MIXER_VOICE_STOPPED) return false;
+  }
+  return true;
+}
+
+bool sts_mixer_stream_stopped(sts_mixer_t* mixer, sts_mixer_stream_t* stream) {
+  for (int i = 0; i < STS_MIXER_VOICES; ++i) {
+    if (mixer->voices[i].stream == stream && mixer->voices[i].state != STS_MIXER_VOICE_STOPPED) return false;
+  }
+  return true;
+}
 
 void sts_mixer_stop_sample(sts_mixer_t* mixer, sts_mixer_sample_t* sample) {
   int i;
@@ -59849,9 +59869,13 @@ void sts_mixer_mix_audio(sts_mixer_t* mixer, void* output, unsigned int samples)
         position = ((int)voice->position) * 2;
         if (position >= voice->stream->sample.length) {
           // buffer empty...refill
-          voice->stream->callback(&voice->stream->sample, voice->stream->userdata);
+          if (voice->stream->callback(&voice->stream->sample, voice->stream->userdata)) {
             voice->position = 0.0f;
             position = 0;
+          } else {
+            sts_mixer__reset_voice(mixer, i);
+            continue;
+          }
         }
         left += sts_mixer__clamp_sample(sts_mixer__get_sample(&voice->stream->sample, position) * voice->gain);
         right += sts_mixer__clamp_sample(sts_mixer__get_sample(&voice->stream->sample, position + 1) * voice->gain);

engine/v4k.c

@@ -4165,6 +4165,7 @@ typedef struct {
     float               dataf[4096*2];
     };
     bool rewind;
+    bool loop;
 } mystream_t;
 
 static void downsample_to_mono_flt( int channels, float *buffer, int samples ) {
@@ -4189,7 +4190,7 @@ static void downsample_to_mono_s16( int channels, short *buffer, int samples ) {
 }
 
 // the callback to refill the (stereo) stream data
-static void refill_stream(sts_mixer_sample_t* sample, void* userdata) {
+static bool refill_stream(sts_mixer_sample_t* sample, void* userdata) {
     mystream_t* stream = (mystream_t*)userdata;
     switch( stream->type ) {
         default:
@@ -4198,6 +4199,7 @@ static void refill_stream(sts_mixer_sample_t* sample, void* userdata) {
             if( stream->rewind ) stream->rewind = 0, ma_dr_wav_seek_to_pcm_frame(&stream->wav, 0);
             if (ma_dr_wav_read_pcm_frames_s16(&stream->wav, sl, (short*)stream->data) < sl) {
                 ma_dr_wav_seek_to_pcm_frame(&stream->wav, 0);
+                if (!stream->loop) return false;
             }
         }
         break; case MP3: {
@@ -4205,6 +4207,7 @@ static void refill_stream(sts_mixer_sample_t* sample, void* userdata) {
             if( stream->rewind ) stream->rewind = 0, ma_dr_mp3_seek_to_pcm_frame(&stream->mp3_, 0);
             if (ma_dr_mp3_read_pcm_frames_f32(&stream->mp3_, sl, stream->dataf) < sl) {
                 ma_dr_mp3_seek_to_pcm_frame(&stream->mp3_, 0);
+                if (!stream->loop) return false;
             }
         }
         break; case OGG: {
@@ -4212,9 +4215,12 @@ static void refill_stream(sts_mixer_sample_t* sample, void* userdata) {
             if( stream->rewind ) stream->rewind = 0, stb_vorbis_seek(stream->ogg, 0);
             if( stb_vorbis_get_samples_short_interleaved(ogg, 2, (short*)stream->data, sample->length) == 0 )  {
                 stb_vorbis_seek(stream->ogg, 0);
+                if (!stream->loop) return false;
             }
         }
     }
+
+    return true;
 }
 static void reset_stream(mystream_t* stream) {
     if( stream ) memset( stream->data, 0, sizeof(stream->data) ), stream->rewind = 1;
@@ -4228,6 +4234,7 @@ static bool load_stream(mystream_t* stream, const char *filename) {
     int error;
     int HZ = 44100;
     stream->type = UNK;
+    stream->loop = true;
     if( stream->type == UNK && (stream->ogg = stb_vorbis_open_memory((const unsigned char *)data, datalen, &error, NULL)) ) {
         stb_vorbis_info info = stb_vorbis_get_info(stream->ogg);
         if( info.channels != 2 ) { puts("cannot stream ogg file. stereo required."); goto end; } // @fixme: upsample
@@ -4539,6 +4546,22 @@ int audio_stop( audio_t a ) {
     return 1;
 }
 
+void audio_loop( audio_t a, bool loop ) {
+    if ( a->is_stream ) {
+        a->stream.loop = loop;
+    }
+}
+
+bool audio_playing( audio_t a ) {
+    if( a->is_clip ) {
+        return !sts_mixer_sample_stopped(&mixer, &a->clip);
+    }
+    if( a->is_stream ) {
+        return !sts_mixer_stream_stopped(&mixer, &a->stream.stream);
+    }
+    return false;
+}
+
 // -----------------------------------------------------------------------------
 // audio queue
 
@@ -4566,7 +4589,7 @@ static void audio_queue_init() {
     do_once thread_queue_init(&queue_mutex, countof(audio_queues), audio_queues, 0);
 }
 
-static void audio_queue_callback(sts_mixer_sample_t* sample, void* userdata) {
+static bool audio_queue_callback(sts_mixer_sample_t* sample, void* userdata) {
     (void)userdata;
 
     int sl = sample->length / 2; // 2 ch
@@ -4590,6 +4613,8 @@ static void audio_queue_callback(sts_mixer_sample_t* sample, void* userdata) {
             aq = 0;
         }
     } while( bytes > 0 );
+
+    return 1;
 }
 
 static int audio_queue_voice = -1;

engine/v4k.h

@@ -1703,6 +1703,8 @@ API int     audio_play_gain( audio_t a, int flags, float gain/*0*/ );
 API int     audio_play_gain_pitch( audio_t a, int flags, float gain, float pitch/*1*/ );
 API int     audio_play_gain_pitch_pan( audio_t a, int flags, float gain, float pitch, float pan/*0*/ );
 API int     audio_stop( audio_t a );
+API void    audio_loop( audio_t a, bool loop );
+API bool    audio_playing( audio_t a );
 
 API float   audio_volume_clip(float gain);   // set     fx volume if gain is in [0..1] range. returns current     fx volume in any case
 API float   audio_volume_stream(float gain); // set    bgm volume if gain is in [0..1] range. returns current    bgm volume in any case

hello.c

@@ -11,8 +11,8 @@
 // - linux/tcc    : tcc hello.c -lm -ldl -lpthread -lX11 -D__STDC_NO_VLA__
 // - osx          : cc -ObjC hello.c -framework cocoa -framework iokit -framework audiotoolbox
 
-#define V4K_IMPLEMENTATION      // unrolls single-header implementation
+// #define V4K_IMPLEMENTATION      // unrolls single-header implementation
-#include "engine/joint/v4k.h"   // single-header file
+#include "engine/v4k.c"   // single-header file
 
 int main() {
     // options

tools/join_3rd.bat

@@ -0,0 +1,3 @@
+#!/bin/bash 2>nul
+
+python tools/join.py --template engine/split/v4k.x.inl --path ./engine/split/ --output ./engine/v4k