remove audio_decoder, cleanup

source/Makefile.mk

@@ -211,19 +211,12 @@ endif
 # --------------------------------------------------------------
 # Check for optional libs (required by internal plugins)
 
-HAVE_AF_DEPS      = $(shell pkg-config --exists sndfile && echo true)
-HAVE_MF_DEPS      = $(shell pkg-config --exists smf && echo true)
-HAVE_PM_DEPS      = $(shell pkg-config --exists libprojectM && echo true)
 HAVE_ZYN_DEPS     = $(shell pkg-config --exists fftw3 mxml zlib && echo true)
 HAVE_ZYN_UI_DEPS  = $(shell pkg-config --exists ntk_images ntk && echo true)
 
 # --------------------------------------------------------------
 # Set base defines
 
-ifeq ($(HAVE_FFMPEG),true)
-BASE_FLAGS += -DHAVE_FFMPEG
-endif
-
 ifeq ($(HAVE_JUCE_UI),true)
 BASE_FLAGS += -DHAVE_JUCE_UI
 endif
@@ -256,15 +249,6 @@ endif
 # --------------------------------------------------------------
 # Set libs stuff (part 2)
 
-ifeq ($(HAVE_AF_DEPS),true)
-AUDIO_DECODER_FLAGS  = $(shell pkg-config --cflags sndfile)
-AUDIO_DECODER_LIBS   = $(shell pkg-config --libs sndfile)
-ifeq ($(HAVE_FFMPEG),true)
-AUDIO_DECODER_FLAGS += $(shell pkg-config --cflags libavcodec libavformat libavutil)
-AUDIO_DECODER_LIBS  += $(shell pkg-config --libs libavcodec libavformat libavutil)
-endif
-endif
-
 RTAUDIO_FLAGS  = -DHAVE_GETTIMEOFDAY -D__UNIX_JACK__
 
 ifeq ($(DEBUG),true)
@@ -334,21 +318,6 @@ endif
 # --------------------------------------------------------------
 # Set libs stuff (part 3)
 
-ifeq ($(HAVE_AF_DEPS),true)
-NATIVE_PLUGINS_FLAGS += -DWANT_AUDIOFILE
-NATIVE_PLUGINS_LIBS  += $(AUDIO_DECODER_LIBS)
-endif
-
-ifeq ($(HAVE_MF_DEPS),true)
-NATIVE_PLUGINS_FLAGS += -DWANT_MIDIFILE
-NATIVE_PLUGINS_LIBS  += $(shell pkg-config --libs smf)
-endif
-
-ifeq ($(HAVE_PM_DEPS),true)
-NATIVE_PLUGINS_FLAGS += -DWANT_PROJECTM
-NATIVE_PLUGINS_LIBS  += $(shell pkg-config --libs libprojectM)
-endif
-
 ifeq ($(HAVE_ZYN_DEPS),true)
 NATIVE_PLUGINS_FLAGS += -DWANT_ZYNADDSUBFX
 NATIVE_PLUGINS_LIBS  += $(shell pkg-config --libs fftw3 mxml zlib)

source/backend/CarlaStandalone.cpp

@@ -593,17 +593,7 @@ const char* carla_get_complete_license_text()
 
 #ifdef WANT_NATIVE
         // Internal plugins
-# ifdef HAVE_OPENGL
-        text3 += "<li>DISTRHO Mini-Series plugin code, based on LOSER-dev suite by Michael Gruhn</li>";
-# endif
         text3 += "<li>NekoFilter plugin code, based on lv2fil by Nedko Arnaudov and Fons Adriaensen</li>";
-        //text1 += "<li>SunVox library file support, http://www.warmplace.ru/soft/sunvox/</li>"; // unfinished
-# ifdef WANT_AUDIOFILE
-        text3 += "<li>AudioDecoder library for Audio file support, by Robin Gareus</li>";
-# endif
-# ifdef WANT_MIDIFILE
-        text3 += "<li>LibSMF library for MIDI file support, http://libsmf.sourceforge.net/</li>";
-# endif
 # ifdef WANT_ZYNADDSUBFX
         text3 += "<li>ZynAddSubFX plugin code, http://zynaddsubfx.sf.net/</li>";
 #  ifdef WANT_ZYNADDSUBFX_UI
@@ -660,16 +650,12 @@ const char* carla_get_supported_file_extensions()
         retText += ";*.gig;*.sfz";
 #endif
 
-        // Files provided by internal plugins
-#ifdef WANT_AUDIOFILE
+        // Audio files, FIXME
         retText += ";*.aiff;*.flac;*.oga;*.ogg;*.w64;*.wav";
-# ifdef HAVE_FFMPEG
         retText += ";*.3g2;*.3gp;*.aac;*.ac3;*.amr;*.ape;*.mp2;*.mp3;*.mpc;*.wma";
-# endif
-#endif
-#ifdef WANT_MIDIFILE
+
+        // MIDI files
         retText += ";*.mid;*.midi";
-#endif
 
         // Plugin presets
 #ifdef WANT_ZYNADDSUBFX

source/backend/Makefile

@@ -25,10 +25,6 @@ ifeq ($(CARLA_PLUGIN_SUPPORT),true)
 STANDALONE_LIBS += ../modules/lilv.a
 endif
 
-ifeq ($(HAVE_AF_DEPS),true)
-STANDALONE_LIBS += ../modules/audio_decoder.a
-endif
-
 ifeq ($(HAVE_JUCE_UI),true)
 STANDALONE_LIBS += ../modules/juce_audio_devices.a
 STANDALONE_LIBS += ../modules/juce_audio_processors.a

source/backend/engine/CarlaEngine.cpp

@@ -913,9 +913,9 @@ bool CarlaEngine::loadFile(const char* const filename)
 
     // -------------------------------------------------------------------
 
+    // FIXME
     if (extension == "aiff" || extension == "flac" || extension == "oga" || extension == "ogg" || extension == "w64" || extension == "wav")
     {
-#ifdef WANT_AUDIOFILE
         if (addPlugin(PLUGIN_INTERNAL, nullptr, baseName, "audiofile", 0, nullptr))
         {
             if (CarlaPlugin* const plugin = getPlugin(pData->curPluginCount-1))
@@ -923,17 +923,11 @@ bool CarlaEngine::loadFile(const char* const filename)
             return true;
         }
         return false;
-#else
-        setLastError("This Carla build does not have Audio file support");
-        return false;
-#endif
     }
 
     if (extension == "3g2" || extension == "3gp" || extension == "aac" || extension == "ac3" || extension == "amr" || extension == "ape" ||
-            extension == "mp2" || extension == "mp3" || extension == "mpc" || extension == "wma")
+        extension == "mp2" || extension == "mp3" || extension == "mpc" || extension == "wma")
     {
-#ifdef WANT_AUDIOFILE
-# ifdef HAVE_FFMPEG
         if (addPlugin(PLUGIN_INTERNAL, nullptr, baseName, "audiofile", 0, nullptr))
         {
             if (CarlaPlugin* const plugin = getPlugin(pData->curPluginCount-1))
@@ -941,21 +935,12 @@ bool CarlaEngine::loadFile(const char* const filename)
             return true;
         }
         return false;
-# else
-        setLastError("This Carla build has Audio file support, but not libav/ffmpeg");
-        return false;
-# endif
-#else
-        setLastError("This Carla build does not have Audio file support");
-        return false;
-#endif
     }
 
     // -------------------------------------------------------------------
 
     if (extension == "mid" || extension == "midi")
     {
-#ifdef WANT_MIDIFILE
         if (addPlugin(PLUGIN_INTERNAL, nullptr, baseName, "midifile", 0, nullptr))
         {
             if (CarlaPlugin* const plugin = getPlugin(pData->curPluginCount-1))
@@ -963,10 +948,6 @@ bool CarlaEngine::loadFile(const char* const filename)
             return true;
         }
         return false;
-#else
-        setLastError("This Carla build does not have MIDI file support");
-        return false;
-#endif
     }
 
     // -------------------------------------------------------------------

source/backend/plugin/NativePlugin.cpp

@@ -2267,21 +2267,6 @@ public:
             pData->iconName = carla_strdup("file");
         else if (std::strcmp(fDescriptor->label, "midifile") == 0)
             pData->iconName = carla_strdup("file");
-        else if (std::strcmp(fDescriptor->label, "sunvoxfile") == 0)
-            pData->iconName = carla_strdup("file");
-
-        else if (std::strcmp(fDescriptor->label, "3BandEQ") == 0)
-            pData->iconName = carla_strdup("distrho");
-        else if (std::strcmp(fDescriptor->label, "3BandSplitter") == 0)
-            pData->iconName = carla_strdup("distrho");
-        else if (std::strcmp(fDescriptor->label, "Nekobi") == 0)
-            pData->iconName = carla_strdup("distrho");
-        else if (std::strcmp(fDescriptor->label, "Notes") == 0)
-            pData->iconName = carla_strdup("distrho");
-        else if (std::strcmp(fDescriptor->label, "PingPongPan") == 0)
-            pData->iconName = carla_strdup("distrho");
-        else if (std::strcmp(fDescriptor->label, "StereoEnhancer") == 0)
-            pData->iconName = carla_strdup("distrho");
 
         // ---------------------------------------------------------------
         // get info

source/bridges/Makefile

@@ -85,14 +85,6 @@ NATIVE_BUILD_FLAGS += $(LINUXSAMPLER_FLAGS)
 NATIVE_LINK_FLAGS  += $(LINUXSAMPLER_LIBS)
 endif
 
-ifeq ($(HAVE_AF_DEPS),true)
-NATIVE_BUILD_FLAGS += -DWANT_AUDIOFILE
-endif
-
-ifeq ($(HAVE_MF_DEPS),true)
-NATIVE_BUILD_FLAGS += -DWANT_MIDIFILE
-endif
-
 ifeq ($(HAVE_ZYN_DEPS),true)
 NATIVE_BUILD_FLAGS += -DWANT_ZYNADDSUBFX
 ifeq ($(HAVE_ZYN_UI_DEPS),true)
@@ -502,11 +494,6 @@ LIBS_NATIVE += \
 	../modules/juce_gui_extra.a
 endif
 
-ifeq ($(HAVE_AF_DEPS),true)
-LIBS_NATIVE += \
-	../modules/audio_decoder.a
-endif
-
 ifeq ($(CARLA_PLUGIN_SUPPORT),true)
 LIBS_NATIVE += \
 	../modules/lilv.a

source/includes/sunvox/sunvox.h

@@ -1,286 +0,0 @@
-#ifndef __SUNVOX_H__
-#define __SUNVOX_H__
-
-#define NOTECMD_NOTE_OFF	128
-#define NOTECMD_ALL_NOTES_OFF	129 /* notes of all synths off */
-#define NOTECMD_CLEAN_SYNTHS	130 /* stop and clean all synths */
-#define NOTECMD_STOP		131
-#define NOTECMD_PLAY		132
-
-//sv_send_event() parameters:
-//  slot;
-//  channel_num: from 0 to 7;
-//  note: 0 - nothing; 1..127 - note num; 128 - note off; 129, 130... - see NOTECMD_xxx defines;
-//  vel: velocity 1..129; 0 - default;
-//  module: 0 - nothing; 1..255 - module number;
-//  ctl: CCXX. CC - number of controller. XX - std effect;
-//  ctl_val: value of controller.
-
-typedef struct
-{
-    unsigned char	note;           //0 - nothing; 1..127 - note num; 128 - note off; 129, 130... - see NOTECMD_xxx defines
-    unsigned char       vel;            //Velocity 1..129; 0 - default
-    unsigned char       module;         //0 - nothing; 1..255 - module number
-    unsigned char       nothing;
-    unsigned short      ctl;            //CCXX. CC - number of controller. XX - std effect
-    unsigned short      ctl_val;        //Value of controller
-} sunvox_note;
-
-#define SV_INIT_FLAG_NO_DEBUG_OUTPUT 		( 1 << 0 )
-#define SV_INIT_FLAG_USER_AUDIO_CALLBACK 	( 1 << 1 ) /* Interaction with sound card is on the user side */
-#define SV_INIT_FLAG_AUDIO_INT16 		( 1 << 2 )
-#define SV_INIT_FLAG_AUDIO_FLOAT32 		( 1 << 3 )
-#define SV_INIT_FLAG_ONE_THREAD			( 1 << 4 ) /* Audio callback and song modification functions are in single thread */
-
-#define SV_MODULE_FLAG_EXISTS 1
-#define SV_MODULE_FLAG_EFFECT 2
-#define SV_MODULE_INPUTS_OFF 16
-#define SV_MODULE_INPUTS_MASK ( 255 << SV_MODULE_INPUTS_OFF )
-#define SV_MODULE_OUTPUTS_OFF ( 16 + 8 )
-#define SV_MODULE_OUTPUTS_MASK ( 255 << SV_MODULE_OUTPUTS_OFF )
-
-#define SV_STYPE_INT16 0
-#define SV_STYPE_INT32 1
-#define SV_STYPE_FLOAT32 2
-#define SV_STYPE_FLOAT64 3
-
-#if defined(_WIN32) || defined(_WIN32_WCE) || defined(__WIN32__)
-    #define WIN
-    #define LIBNAME "sunvox.dll"
-#endif
-#if defined(__APPLE__) 
-    #define OSX
-    #define LIBNAME "sunvox.dylib"
-#endif
-#if defined(__linux__) || defined(linux)
-    #define LINUX
-    #define LIBNAME "sunvox.so"
-#endif
-#if defined(OSX) || defined(LINUX)
-    #define UNIX
-#endif
-
-#ifdef WIN
-    #define SUNVOX_FN_ATTR __attribute__((stdcall))
-#endif
-#ifndef SUNVOX_FN_ATTR
-    #define SUNVOX_FN_ATTR /**/
-#endif
-
-//sv_audio_callback() - get the next piece of SunVox audio.
-//buf - destination buffer of type signed short (if SV_INIT_FLAG_AUDIO_INT16 used in sv_init())
-//      or float (if SV_INIT_FLAG_AUDIO_FLOAT32 used in sv_init());
-//	stereo data will be interleaved in this buffer: LRLR... ; where the LR is the one frame;
-//frames - number of frames in destination buffer;
-//latency - audio latency (in frames);
-//out_time - output time (in ticks).
-typedef int (*tsv_audio_callback)( void* buf, int frames, int latency, unsigned int out_time ) SUNVOX_FN_ATTR;
-typedef int (*tsv_open_slot)( int slot ) SUNVOX_FN_ATTR;
-typedef int (*tsv_close_slot)( int slot ) SUNVOX_FN_ATTR;
-typedef int (*tsv_lock_slot)( int slot ) SUNVOX_FN_ATTR;
-typedef int (*tsv_unlock_slot)( int slot ) SUNVOX_FN_ATTR;
-typedef int (*tsv_init)( const char* dev, int freq, int channels, int flags ) SUNVOX_FN_ATTR;
-typedef int (*tsv_deinit)( void ) SUNVOX_FN_ATTR;
-//sv_get_sample_type() - get internal sample type of the SunVox engine. Return value: one of the SV_STYPE_xxx defines.
-//Use it to get the scope buffer type from get_module_scope() function.
-typedef int (*tsv_get_sample_type)( void ) SUNVOX_FN_ATTR;
-typedef int (*tsv_load)( int slot, const char* name ) SUNVOX_FN_ATTR;
-typedef int (*tsv_load_from_memory)( int slot, void* data, unsigned int data_size ) SUNVOX_FN_ATTR;
-typedef int (*tsv_play)( int slot ) SUNVOX_FN_ATTR;
-typedef int (*tsv_play_from_beginning)( int slot ) SUNVOX_FN_ATTR;
-typedef int (*tsv_stop)( int slot ) SUNVOX_FN_ATTR;
-//autostop values: 0 - disable autostop; 1 - enable autostop.
-//When disabled, song is playing infinitely in the loop.
-typedef int (*tsv_set_autostop)( int slot, int autostop ) SUNVOX_FN_ATTR;
-//sv_end_of_song() return values: 0 - song is playing now; 1 - stopped.
-typedef int (*tsv_end_of_song)( int slot ) SUNVOX_FN_ATTR;
-typedef int (*tsv_rewind)( int slot, int t ) SUNVOX_FN_ATTR;
-typedef int (*tsv_volume)( int slot, int vol ) SUNVOX_FN_ATTR;
-typedef int (*tsv_send_event)( int slot, int channel_num, int note, int vel, int module, int ctl, int ctl_val ) SUNVOX_FN_ATTR;
-typedef int (*tsv_get_current_line)( int slot ) SUNVOX_FN_ATTR;
-typedef int (*tsv_get_current_signal_level)( int slot, int channel ) SUNVOX_FN_ATTR; //From 0 to 255
-typedef const char* (*tsv_get_song_name)( int slot ) SUNVOX_FN_ATTR;
-typedef int (*tsv_get_song_bpm)( int slot ) SUNVOX_FN_ATTR;
-typedef int (*tsv_get_song_tpl)( int slot ) SUNVOX_FN_ATTR;
-//Frame is one discrete of the sound. Sampling frequency 44100 Hz means, that you hear 44100 frames per second.
-typedef unsigned int (*tsv_get_song_length_frames)( int slot ) SUNVOX_FN_ATTR;
-typedef unsigned int (*tsv_get_song_length_lines)( int slot ) SUNVOX_FN_ATTR;
-typedef int (*tsv_get_number_of_modules)( int slot ) SUNVOX_FN_ATTR;
-typedef int (*tsv_get_module_flags)( int slot, int mod_num ) SUNVOX_FN_ATTR;
-typedef int* (*tsv_get_module_inputs)( int slot, int mod_num ) SUNVOX_FN_ATTR;
-typedef int* (*tsv_get_module_outputs)( int slot, int mod_num ) SUNVOX_FN_ATTR;
-typedef const char* (*tsv_get_module_name)( int slot, int mod_num ) SUNVOX_FN_ATTR;
-typedef unsigned int (*tsv_get_module_xy)( int slot, int mod_num ) SUNVOX_FN_ATTR;
-typedef int (*tsv_get_module_color)( int slot, int mod_num ) SUNVOX_FN_ATTR;
-typedef void* (*tsv_get_module_scope)( int slot, int mod_num, int channel, int* offset, int* buffer_size ) SUNVOX_FN_ATTR;
-typedef int (*tsv_get_number_of_patterns)( int slot ) SUNVOX_FN_ATTR;
-typedef int (*tsv_get_pattern_x)( int slot, int pat_num ) SUNVOX_FN_ATTR;
-typedef int (*tsv_get_pattern_y)( int slot, int pat_num ) SUNVOX_FN_ATTR;
-typedef int (*tsv_get_pattern_tracks)( int slot, int pat_num ) SUNVOX_FN_ATTR;
-typedef int (*tsv_get_pattern_lines)( int slot, int pat_num ) SUNVOX_FN_ATTR;
-typedef sunvox_note* (*tsv_get_pattern_data)( int slot, int pat_num ) SUNVOX_FN_ATTR;
-typedef int (*tsv_pattern_mute)( int slot, int pat_num, int mute ) SUNVOX_FN_ATTR; //Use it with sv_lock_slot() and sv_unlock_slot()
-//SunVox engine uses its own time space, measured in ticks.
-//Use sv_get_ticks() to get current tick counter (from 0 to 0xFFFFFFFF).
-//Use sv_get_ticks_per_second() to get the number of SunVox ticks per second.
-typedef unsigned int (*tsv_get_ticks)( void ) SUNVOX_FN_ATTR;
-typedef unsigned int (*tsv_get_ticks_per_second)( void ) SUNVOX_FN_ATTR;
-
-#ifdef SUNVOX_MAIN
-
-#ifdef WIN
-#define IMPORT( Handle, Type, Function, Store ) \
-    { \
-	Store = (Type)GetProcAddress( Handle, Function ); \
-	if( Store == 0 ) { fn_not_found = Function; break; } \
-    }
-#define ERROR_MSG( msg ) MessageBox( 0, TEXT("msg"), TEXT("Error"), MB_OK );
-#endif
-
-#ifdef UNIX
-#define IMPORT( Handle, Type, Function, Store ) \
-    { \
-	Store = (Type)dlsym( Handle, Function ); \
-	if( Store == 0 ) { fn_not_found = Function; break; } \
-    }
-#define ERROR_MSG( msg ) printf( "ERROR: %s\n", msg );
-#endif
-
-tsv_audio_callback sv_audio_callback = 0;
-tsv_open_slot sv_open_slot = 0;
-tsv_close_slot sv_close_slot = 0;
-tsv_lock_slot sv_lock_slot = 0;
-tsv_unlock_slot sv_unlock_slot = 0;
-tsv_init sv_init = 0;
-tsv_deinit sv_deinit = 0;
-tsv_get_sample_type sv_get_sample_type = 0;
-tsv_load sv_load = 0;
-tsv_load_from_memory sv_load_from_memory = 0;
-tsv_play sv_play = 0;
-tsv_play_from_beginning sv_play_from_beginning = 0;
-tsv_stop sv_stop = 0;
-tsv_set_autostop sv_set_autostop = 0;
-tsv_end_of_song sv_end_of_song = 0;
-tsv_rewind sv_rewind = 0;
-tsv_volume sv_volume = 0;
-tsv_send_event sv_send_event = 0;
-tsv_get_current_line sv_get_current_line = 0;
-tsv_get_current_signal_level sv_get_current_signal_level = 0;
-tsv_get_song_name sv_get_song_name = 0;
-tsv_get_song_bpm sv_get_song_bpm = 0;
-tsv_get_song_tpl sv_get_song_tpl = 0;
-tsv_get_song_length_frames sv_get_song_length_frames = 0;
-tsv_get_song_length_lines sv_get_song_length_lines = 0;
-tsv_get_number_of_modules sv_get_number_of_modules = 0;
-tsv_get_module_flags sv_get_module_flags = 0;
-tsv_get_module_inputs sv_get_module_inputs = 0;
-tsv_get_module_outputs sv_get_module_outputs = 0;
-tsv_get_module_name sv_get_module_name = 0;
-tsv_get_module_xy sv_get_module_xy = 0;
-tsv_get_module_color sv_get_module_color = 0;
-tsv_get_module_scope sv_get_module_scope = 0;
-tsv_get_number_of_patterns sv_get_number_of_patterns = 0;
-tsv_get_pattern_x sv_get_pattern_x = 0;
-tsv_get_pattern_y sv_get_pattern_y = 0;
-tsv_get_pattern_tracks sv_get_pattern_tracks = 0;
-tsv_get_pattern_lines sv_get_pattern_lines = 0;
-tsv_get_pattern_data sv_get_pattern_data = 0;
-tsv_pattern_mute sv_pattern_mute = 0;
-tsv_get_ticks sv_get_ticks = 0;
-tsv_get_ticks_per_second sv_get_ticks_per_second = 0;
-
-#ifdef UNIX
-    void* g_sv_dll = 0;
-#endif
-
-#ifdef WIN
-    HMODULE g_sv_dll = 0;
-#endif
-
-int sv_load_dll( void )
-{
-#ifdef WIN
-    g_sv_dll = LoadLibrary( TEXT(LIBNAME) );
-    if( g_sv_dll == 0 ) 
-    {
-        ERROR_MSG( "sunvox.dll not found" );
-        return 1;
-    }
-#endif
-#ifdef UNIX
-    g_sv_dll = dlopen( LIBNAME, RTLD_NOW );
-    if( g_sv_dll == 0 )
-    {
-	printf( "%s\n", dlerror() );
-        return 1;
-    }
-#endif
-    const char* fn_not_found = 0;
-    while( 1 )
-    {
-	IMPORT( g_sv_dll, tsv_audio_callback, "sv_audio_callback", sv_audio_callback );
-	IMPORT( g_sv_dll, tsv_open_slot, "sv_open_slot", sv_open_slot );
-	IMPORT( g_sv_dll, tsv_close_slot, "sv_close_slot", sv_close_slot );
-	IMPORT( g_sv_dll, tsv_lock_slot, "sv_lock_slot", sv_lock_slot );
-	IMPORT( g_sv_dll, tsv_unlock_slot, "sv_unlock_slot", sv_unlock_slot );
-	IMPORT( g_sv_dll, tsv_init, "sv_init", sv_init );
-	IMPORT( g_sv_dll, tsv_deinit, "sv_deinit", sv_deinit );
-	IMPORT( g_sv_dll, tsv_get_sample_type, "sv_get_sample_type", sv_get_sample_type );
-	IMPORT( g_sv_dll, tsv_load, "sv_load", sv_load );
-	IMPORT( g_sv_dll, tsv_load_from_memory, "sv_load_from_memory", sv_load_from_memory );
-	IMPORT( g_sv_dll, tsv_play, "sv_play", sv_play );
-	IMPORT( g_sv_dll, tsv_play_from_beginning, "sv_play_from_beginning", sv_play_from_beginning );
-	IMPORT( g_sv_dll, tsv_stop, "sv_stop", sv_stop );
-	IMPORT( g_sv_dll, tsv_set_autostop, "sv_set_autostop", sv_set_autostop );
-	IMPORT( g_sv_dll, tsv_end_of_song, "sv_end_of_song", sv_end_of_song );
-	IMPORT( g_sv_dll, tsv_rewind, "sv_rewind", sv_rewind );
-	IMPORT( g_sv_dll, tsv_volume, "sv_volume", sv_volume );
-	IMPORT( g_sv_dll, tsv_send_event, "sv_send_event", sv_send_event );
-	IMPORT( g_sv_dll, tsv_get_current_line, "sv_get_current_line", sv_get_current_line );
-	IMPORT( g_sv_dll, tsv_get_current_signal_level, "sv_get_current_signal_level", sv_get_current_signal_level );
-	IMPORT( g_sv_dll, tsv_get_song_name, "sv_get_song_name", sv_get_song_name );
-	IMPORT( g_sv_dll, tsv_get_song_bpm, "sv_get_song_bpm", sv_get_song_bpm );
-	IMPORT( g_sv_dll, tsv_get_song_tpl, "sv_get_song_tpl", sv_get_song_tpl );
-	IMPORT( g_sv_dll, tsv_get_song_length_frames, "sv_get_song_length_frames", sv_get_song_length_frames );
-	IMPORT( g_sv_dll, tsv_get_song_length_lines, "sv_get_song_length_lines", sv_get_song_length_lines );
-	IMPORT( g_sv_dll, tsv_get_number_of_modules, "sv_get_number_of_modules", sv_get_number_of_modules );
-	IMPORT( g_sv_dll, tsv_get_module_flags, "sv_get_module_flags", sv_get_module_flags );
-	IMPORT( g_sv_dll, tsv_get_module_inputs, "sv_get_module_inputs", sv_get_module_inputs );
-	IMPORT( g_sv_dll, tsv_get_module_outputs, "sv_get_module_outputs", sv_get_module_outputs );
-	IMPORT( g_sv_dll, tsv_get_module_name, "sv_get_module_name", sv_get_module_name );
-	IMPORT( g_sv_dll, tsv_get_module_xy, "sv_get_module_xy", sv_get_module_xy );
-	IMPORT( g_sv_dll, tsv_get_module_color, "sv_get_module_color", sv_get_module_color );
-	IMPORT( g_sv_dll, tsv_get_module_scope, "sv_get_module_scope", sv_get_module_scope );
-	IMPORT( g_sv_dll, tsv_get_number_of_patterns, "sv_get_number_of_patterns", sv_get_number_of_patterns );
-	IMPORT( g_sv_dll, tsv_get_pattern_x, "sv_get_pattern_x", sv_get_pattern_x );
-	IMPORT( g_sv_dll, tsv_get_pattern_y, "sv_get_pattern_y", sv_get_pattern_y );
-	IMPORT( g_sv_dll, tsv_get_pattern_tracks, "sv_get_pattern_tracks", sv_get_pattern_tracks );
-	IMPORT( g_sv_dll, tsv_get_pattern_lines, "sv_get_pattern_lines", sv_get_pattern_lines );
-	IMPORT( g_sv_dll, tsv_get_pattern_data, "sv_get_pattern_data", sv_get_pattern_data );
-	IMPORT( g_sv_dll, tsv_pattern_mute, "sv_pattern_mute", sv_pattern_mute );
-	IMPORT( g_sv_dll, tsv_get_ticks, "sv_get_ticks", sv_get_ticks );
-	IMPORT( g_sv_dll, tsv_get_ticks_per_second, "sv_get_ticks_per_second", sv_get_ticks_per_second );
-	break;
-    }
-    if( fn_not_found )
-    {
-	char ts[ 256 ];
-	sprintf( ts, "sunvox lib: %s() not found", fn_not_found );
-	ERROR_MSG( ts );
-	return -1;
-    }
-    
-    return 0;
-}
-
-int sv_unload_dll( void )
-{
-#ifdef UNIX
-    if( g_sv_dll ) dlclose( g_sv_dll );
-#endif
-    return 0;
-}
-
-#endif
-
-#endif

source/modules/Makefile

@@ -60,8 +60,6 @@ jackbridge-wine64:
 
 clean:
 	rm -f *.a *.def *.dll *.dylib *.so
-	$(MAKE) clean -C audio_decoder
-	$(MAKE) clean -C dgl
 	$(MAKE) clean -C jackbridge
 	$(MAKE) clean -C juce_audio_basics
 	$(MAKE) clean -C juce_audio_devices
@@ -78,13 +76,12 @@ clean:
 	$(MAKE) clean -C rtaudio
 	$(MAKE) clean -C rtmempool
 	$(MAKE) clean -C rtmidi
-	$(MAKE) clean -C stk
 	$(MAKE) clean -C theme
 
 # --------------------------------------------------------------
 
 .PHONY: \
-	audio_decoder dgl jackbridge lilv native-plugins rtaudio rtmempool rtmidi stk theme \
+	jackbridge lilv native-plugins rtaudio rtmempool rtmidi theme \
 	juce_audio_basics juce_audio_devices juce_audio_formats juce_audio_processors juce_core juce_data_structures juce_events juce_graphics juce_gui_basics juce_gui_extra
 
 # --------------------------------------------------------------

source/modules/audio_decoder/Makefile

@@ -1,53 +0,0 @@
-#!/usr/bin/make -f
-# Makefile for audio_decoder #
-# -------------------------- #
-# Created by falkTX
-#
-
-include ../../Makefile.mk
-
-# --------------------------------------------------------------
-
-BUILD_C_FLAGS += $(AUDIO_DECODER_FLAGS) -I. -I.. -w
-
-# --------------------------------------------------------------
-
-OBJS = \
-	ad_ffmpeg.c.o \
-	ad_plugin.c.o \
-	ad_soundfile.c.o
-
-TARGETS = \
-	../audio_decoder.a
-
-# --------------------------------------------------------------
-
-all: $(TARGETS)
-
-# --------------------------------------------------------------
-
-clean:
-	$(RM) $(OBJS)
-	$(RM) $(TARGETS)
-
-debug:
-	$(MAKE) DEBUG=true
-
-# --------------------------------------------------------------
-
-../audio_decoder.a: $(OBJS)
-	$(RM) $@
-	$(AR) crs $@ $^
-
-# --------------------------------------------------------------
-
-ad_ffmpeg.c.o: ad_ffmpeg.c ad_plugin.h ffcompat.h ad.h
-	$(CC) $< $(BUILD_C_FLAGS) -c -o $@
-
-ad_plugin.c.o: ad_plugin.c ad_plugin.h ad.h
-	$(CC) $< $(BUILD_C_FLAGS) -c -o $@
-
-ad_soundfile.c.o: ad_soundfile.c ad_plugin.h ad.h
-	$(CC) $< $(BUILD_C_FLAGS) -c -o $@
-
-# --------------------------------------------------------------

source/modules/audio_decoder/ad.h

@@ -1,127 +0,0 @@
-/**
-   @brief audio-decoder - wrapper around libsndfile and libav*
-   @file ad.h
-   @author Robin Gareus <robin@gareus.org>
-
-   Copyright (C) 2011-2013 Robin Gareus <robin@gareus.org>
-
-   This program is free software; you can redistribute it and/or modify
-   it under the terms of the GNU Lesser Public License as published by
-   the Free Software Foundation; either version 2.1, or (at your option)
-   any later version.
-
-   This program is distributed in the hope that it will be useful,
-   but WITHOUT ANY WARRANTY; without even the implied warranty of
-   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-   GNU Lesser Public License for more details.
-
-   You should have received a copy of the GNU Lesser General Public
-   License along with this library; if not, write to the Free Software
-   Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-*/
-#ifndef __AD_H__
-#define __AD_H__
-
-#include <unistd.h>
-#include <stdint.h>
-
-struct adinfo {
-	unsigned int sample_rate;
-	unsigned int channels;
-	int64_t length; //milliseconds
-	int64_t frames; //total number of frames (eg a frame for 16bit stereo is 4 bytes).
-	int     bit_rate;
-	int     bit_depth;
-	char *  meta_data;
-};
-
-/* global init function - register codecs */
-void ad_init();
-
-/* --- public API --- */
-
-/** open an audio file
- * @param fn file-name
- * @param nfo pointer to a adinfo struct which will hold information about the file.
- * @return NULL on error, a pointer to an opaque soundfile-decoder object on success.
- */
-void *  ad_open  (const char *fn, struct adinfo *nfo);
-
-/** close an audio file and release decoder structures
- * @param sf decoder handle
- * @return 0 on succees, -1 if sf was invalid or not open (return value can usually be ignored)
- */
-int     ad_close (void *sf);
-
-/** seel to a given position in the file
- * @param sf decoder handle
- * @param pos frame position to seek to in frames (1 frame = number-of-channel samples) from the start of the file.
- * @return the current position in frames (multi-channel samples) from the start of the file. On error this function returns -1.
- */
-int64_t ad_seek  (void *sf, int64_t pos);
-
-/** decode audio data chunk to raw interleaved channel floating point data
- *
- * @param sf decoder handle
- * @param out place to store data -- must be large enough to hold  (sizeof(float) * len) bytes.
- * @param len number of samples (!) to read (should be a multiple of nfo->channels).
- * @return the number of read samples.
- */
-ssize_t ad_read  (void *sf, float* out, size_t len);
-
-/** re-read the file information and meta-data.
- *
- * this is not neccesary in general \ref ad_open includes an inplicit call
- * but meta-data may change in live-stream in which case en explicit call to
- * ad_into is needed to update the inforation
- *
- * @param fn file-name
- * @param nfo pointer to a adinfo struct which will hold information about the file.
- * @return 0 on succees, -1 if sf was invalid or not open
- */
-int     ad_info  (void *sf, struct adinfo *nfo);
-
-/** zero initialize the information struct.  * (does not free nfo->meta_data)
- * @param nfo pointer to a adinfo struct
- */
-void    ad_clear_nfo     (struct adinfo *nfo);
-
-/** free possibly allocated meta-data text
- * @param nfo pointer to a adinfo struct
- */
-void    ad_free_nfo      (struct adinfo *nfo);
-
-
-/* --- helper functions --- */
-
-/** read file info
- *  combines ad_open() and ad_close()
- */
-int ad_finfo             (const char *, struct adinfo *);
-
-/**
- * wrapper around \ref ad_read, downmixes all channels to mono
- */
-ssize_t ad_read_mono_dbl (void *, struct adinfo *, double*, size_t);
-
-/**
- * calls dbg() to print file info to stderr.
- *
- * @param dbglvl
- * @param nfo
- */
-void ad_dump_nfo            (int dbglvl, struct adinfo *nfo);
-
-/** set audio-decoder debug level  -- all info is printed to stderr.
- *
- * @param lvl debug-level threshold
- * -1: absolutley silent
- *  0: errors only
- *  1: errors + info
- *  2: + debug
- *  3: + low-level-debug info
- */
-void ad_set_debuglevel(int lvl);
-
-#endif

source/modules/audio_decoder/ad_ffmpeg.c

@@ -1,373 +0,0 @@
-/**
-   Copyright (C) 2011-2013 Robin Gareus <robin@gareus.org>
-
-   This program is free software; you can redistribute it and/or modify
-   it under the terms of the GNU Lesser Public License as published by
-   the Free Software Foundation; either version 2.1, or (at your option)
-   any later version.
-
-   This program is distributed in the hope that it will be useful,
-   but WITHOUT ANY WARRANTY; without even the implied warranty of
-   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-   GNU Lesser Public License for more details.
-
-   You should have received a copy of the GNU Lesser General Public
-   License along with this library; if not, write to the Free Software
-   Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-*/
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <unistd.h>
-#include <math.h>
-
-#include "ad_plugin.h"
-
-#ifdef HAVE_FFMPEG
-
-#include "ffcompat.h"
-
-#ifndef MIN
-#define MIN(a,b) ( ( (a) < (b) )? (a) : (b) )
-#endif
-
-typedef struct {
-  AVFormatContext* formatContext;
-  AVCodecContext*  codecContext;
-  AVCodec*         codec;
-  AVPacket         packet;
-  int              audioStream;
-  int              pkt_len;
-  uint8_t*         pkt_ptr;
-
-  int16_t          m_tmpBuffer[AVCODEC_MAX_AUDIO_FRAME_SIZE];
-  int16_t*         m_tmpBufferStart;
-  unsigned long    m_tmpBufferLen;
-
-  int64_t          decoder_clock;
-  int64_t          output_clock;
-  int64_t          seek_frame;
-  unsigned int     samplerate;
-  unsigned int     channels;
-  int64_t          length;
-} ffmpeg_audio_decoder;
-
-
-static int ad_info_ffmpeg(void *sf, struct adinfo *nfo) {
-  ffmpeg_audio_decoder *priv = (ffmpeg_audio_decoder*) sf;
-  if (!priv) return -1;
-  if (nfo) {
-    nfo->sample_rate = priv->samplerate;
-    nfo->channels    = priv->channels;
-    nfo->frames      = priv->length;
-    if (nfo->sample_rate==0) return -1;
-    nfo->length      = (nfo->frames * 1000) / nfo->sample_rate;
-    nfo->bit_rate    = priv->formatContext->bit_rate;
-    nfo->bit_depth   = 0;
-    nfo->meta_data   = NULL;
-
-#ifdef WITH_GTK // XXX replace g_* functions with POSIX equiv
-		AVDictionaryEntry *tag = NULL;
-		// Tags in container
-		while ((tag = av_dict_get(priv->formatContext->metadata, "", tag, AV_DICT_IGNORE_SUFFIX))) {
-			dbg(2, "FTAG: %s=%s", tag->key, tag->value);
-			char * tmp = g_strdup_printf("%s%s<i>%s</i>:%s", nfo->meta_data?nfo->meta_data:"",nfo->meta_data?"\n":"", tag->key, tag->value);
-			if (nfo->meta_data) g_free(nfo->meta_data);
-			nfo->meta_data = tmp;
-		}
-		// Tags in stream
-		tag=NULL;
-		AVStream *stream = priv->formatContext->streams[priv->audioStream];
-		while ((tag = av_dict_get(stream->metadata, "", tag, AV_DICT_IGNORE_SUFFIX))) {
-			dbg(2, "STAG: %s=%s", tag->key, tag->value);
-			char * tmp = g_strdup_printf("%s%s<i>%s</i>:%s", nfo->meta_data?nfo->meta_data:"",nfo->meta_data?"\n":"", tag->key, tag->value);
-			if (nfo->meta_data) g_free(nfo->meta_data);
-			nfo->meta_data = tmp;
-		}
-#endif
-  }
-  return 0;
-}
-
-static void *ad_open_ffmpeg(const char *fn, struct adinfo *nfo) {
-  ffmpeg_audio_decoder *priv = (ffmpeg_audio_decoder*) calloc(1, sizeof(ffmpeg_audio_decoder));
-  
-  priv->m_tmpBufferStart=NULL;
-  priv->m_tmpBufferLen=0;
-  priv->decoder_clock=priv->output_clock=priv->seek_frame=0; 
-  priv->packet.size=0; priv->packet.data=NULL;
-
-  if (avformat_open_input(&priv->formatContext, fn, NULL, NULL) <0) {
-    dbg(0, "ffmpeg is unable to open file '%s'.", fn);
-    free(priv); return(NULL);
-  }
-
-  if (avformat_find_stream_info(priv->formatContext, NULL) < 0) {
-    avformat_close_input(&priv->formatContext);
-    dbg(0, "av_find_stream_info failed" );
-    free(priv); return(NULL);
-  }
-
-  priv->audioStream = -1;
-  unsigned int i;
-  for (i=0; i<priv->formatContext->nb_streams; i++) {
-    if (priv->formatContext->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
-      priv->audioStream = i;
-      break;
-    }
-  }
-  if (priv->audioStream == -1) {
-    dbg(0, "No Audio Stream found in file");
-    avformat_close_input(&priv->formatContext);
-    free(priv); return(NULL);
-  }
-
-  priv->codecContext = priv->formatContext->streams[priv->audioStream]->codec;
-  priv->codec        = avcodec_find_decoder(priv->codecContext->codec_id);
-
-  if (priv->codec == NULL) {
-    avformat_close_input(&priv->formatContext);
-    dbg(0, "Codec not supported by ffmpeg");
-    free(priv); return(NULL);
-  }
-  if (avcodec_open2(priv->codecContext, priv->codec, NULL) < 0) {
-    dbg(0, "avcodec_open failed" );
-    free(priv); return(NULL);
-  }
-
-  dbg(2, "ffmpeg - audio tics: %i/%i [sec]",priv->formatContext->streams[priv->audioStream]->time_base.num,priv->formatContext->streams[priv->audioStream]->time_base.den);
-
-  int64_t len = priv->formatContext->duration - priv->formatContext->start_time;
-
-  priv->formatContext->flags|=AVFMT_FLAG_GENPTS;
-  priv->formatContext->flags|=AVFMT_FLAG_IGNIDX;
-
-  priv->samplerate = priv->codecContext->sample_rate;
-  priv->channels   = priv->codecContext->channels ;
-  priv->length     = (int64_t)( len * priv->samplerate / AV_TIME_BASE );
-
-  if (ad_info_ffmpeg((void*)priv, nfo)) {
-    dbg(0, "invalid file info (sample-rate==0)");
-    free(priv); return(NULL);
-  }
-
-  dbg(1, "ffmpeg - %s", fn);
-  if (nfo) 
-    dbg(1, "ffmpeg - sr:%i c:%i d:%"PRIi64" f:%"PRIi64, nfo->sample_rate, nfo->channels, nfo->length, nfo->frames);
-
-  return (void*) priv;
-}
-
-static int ad_close_ffmpeg(void *sf) {
-  ffmpeg_audio_decoder *priv = (ffmpeg_audio_decoder*) sf;
-  if (!priv) return -1;
-  avcodec_close(priv->codecContext);
-  avformat_close_input(&priv->formatContext);
-  free(priv);
-  return 0;
-}
-
-static void int16_to_float(int16_t *in, float *out, int num_channels, int num_samples, int out_offset) {
-  int i,ii;
-  for (i=0;i<num_samples;i++) {
-    for (ii=0;ii<num_channels;ii++) {
-     out[(i+out_offset)*num_channels+ii]= (float) in[i*num_channels+ii]/ 32768.0;
-    }
-  }
-}
-
-static ssize_t ad_read_ffmpeg(void *sf, float* d, size_t len) {
-  ffmpeg_audio_decoder *priv = (ffmpeg_audio_decoder*) sf;
-  if (!priv) return -1;
-  size_t frames = len / priv->channels;
-
-  size_t written = 0;
-  ssize_t ret = 0;
-  while (ret >= 0 && written < frames) {
-    dbg(3,"loop: %i/%i (bl:%lu)",written, frames, priv->m_tmpBufferLen );
-    if (priv->seek_frame == 0 && priv->m_tmpBufferLen > 0 ) {
-      int s = MIN(priv->m_tmpBufferLen / priv->channels, frames - written );
-      int16_to_float(priv->m_tmpBufferStart, d, priv->channels, s , written);
-      written += s;
-      priv->output_clock+=s;
-      s = s * priv->channels;
-      priv->m_tmpBufferStart += s;
-      priv->m_tmpBufferLen -= s;
-      ret = 0;
-    } else {
-      priv->m_tmpBufferStart = priv->m_tmpBuffer;
-      priv->m_tmpBufferLen = 0;
-
-      if (!priv->pkt_ptr || priv->pkt_len <1 ) {
-        if (priv->packet.data) av_free_packet(&priv->packet);
-        ret = av_read_frame(priv->formatContext, &priv->packet);
-        if (ret<0) { dbg(1, "reached end of file."); break; }
-        priv->pkt_len = priv->packet.size;
-        priv->pkt_ptr = priv->packet.data;
-      }
-
-      if (priv->packet.stream_index != priv->audioStream) {
-        priv->pkt_ptr = NULL;
-        continue;
-      }
-
-      /* decode all chunks in packet */
-      int data_size= AVCODEC_MAX_AUDIO_FRAME_SIZE;
-
-#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 0, 0)
-      /* This works but is not optimal (channels may not be planar/interleaved) */
-      AVFrame avf; // TODO statically allocate as poart of priv->..
-      memset(&avf, 0, sizeof(AVFrame)); // not sure if that is needed
-      int got_frame = 0;
-      ret = avcodec_decode_audio4(priv->codecContext, &avf, &got_frame, &priv->packet);
-      data_size = avf.linesize[0];
-      memcpy(priv->m_tmpBuffer, avf.data[0], avf.linesize[0] * sizeof(uint8_t));
-#elif LIBAVUTIL_VERSION_INT > AV_VERSION_INT(49, 15, 0) && LIBAVCODEC_VERSION_INT > AV_VERSION_INT(52, 20, 1) // ??
-      // this was deprecated in LIBAVCODEC_VERSION_MAJOR 53
-      ret = avcodec_decode_audio3(priv->codecContext,
-          priv->m_tmpBuffer, &data_size, &priv->packet);
-#else
-			int len = priv->packet.size;
-			uint8_t *ptr = priv->packet.data;
-      ret = avcodec_decode_audio2(priv->codecContext,
-          priv->m_tmpBuffer, &data_size, ptr, len);
-#endif
-
-      if (ret < 0 || ret > priv->pkt_len) {
-#if 0
-        dbg(0, "audio decode error");
-        return -1;
-#endif
-        priv->pkt_len=0;
-        ret=0;
-        continue;
-      }
-
-      priv->pkt_len -= ret; priv->pkt_ptr += ret;
-
-      /* sample exact alignment  */
-      if (priv->packet.pts != AV_NOPTS_VALUE) {
-        priv->decoder_clock = priv->samplerate * av_q2d(priv->formatContext->streams[priv->audioStream]->time_base) * priv->packet.pts;
-      } else {
-        dbg(0, "!!! NO PTS timestamp in file");
-        priv->decoder_clock += (data_size>>1) / priv->channels;
-      }
-
-      if (data_size>0) {
-        priv->m_tmpBufferLen+= (data_size>>1); // 2 bytes per sample
-      }
-
-      /* align buffer after seek. */
-      if (priv->seek_frame > 0) { 
-        const int diff = priv->output_clock-priv->decoder_clock;
-        if (diff<0) { 
-          /* seek ended up past the wanted sample */
-          dbg(0, " !!! Audio seek failed.");
-          return -1;
-        } else if (priv->m_tmpBufferLen < (diff*priv->channels)) {
-          /* wanted sample not in current buffer - keep going */
-          dbg(2, " !!! seeked sample was not in decoded buffer. frames-to-go: %li", diff);
-          priv->m_tmpBufferLen = 0;
-        } else if (diff!=0 && data_size > 0) {
-          /* wanted sample is in current buffer but not at the beginnning */
-          dbg(2, " !!! sync buffer to seek. (diff:%i)", diff);
-          priv->m_tmpBufferStart+= diff*priv->codecContext->channels;
-          priv->m_tmpBufferLen  -= diff*priv->codecContext->channels;
-#if 1
-          memmove(priv->m_tmpBuffer, priv->m_tmpBufferStart, priv->m_tmpBufferLen);
-          priv->m_tmpBufferStart = priv->m_tmpBuffer;
-#endif
-          priv->seek_frame=0;
-          priv->decoder_clock += diff;
-        } else if (data_size > 0) {
-          dbg(2, "Audio exact sync-seek (%"PRIi64" == %"PRIi64")", priv->decoder_clock, priv->seek_frame);
-          priv->seek_frame=0;
-        } else {
-          dbg(0, "Error: no audio data in packet");
-        }
-      }
-      //dbg(0, "PTS: decoder:%"PRIi64". - want: %"PRIi64, priv->decoder_clock, priv->output_clock);
-      //dbg(0, "CLK: frame:  %"PRIi64"  T:%.3fs",priv->decoder_clock, (float) priv->decoder_clock/priv->samplerate);
-    }
-  }
-  if (written!=frames) {
-        dbg(2, "short-read");
-  }
-  return written * priv->channels;
-}
-
-static int64_t ad_seek_ffmpeg(void *sf, int64_t pos) {
-  ffmpeg_audio_decoder *priv = (ffmpeg_audio_decoder*) sf;
-  if (!sf) return -1;
-  if (pos == priv->output_clock) return pos;
-
-  /* flush internal buffer */
-  priv->m_tmpBufferLen = 0;
-  priv->seek_frame = pos;
-  priv->output_clock = pos;
-  priv->pkt_len = 0; priv->pkt_ptr = NULL;
-  priv->decoder_clock = 0;
-
-#if 0
-  /* TODO seek at least 1 packet before target.
-   * for mpeg compressed files, the
-   * output may depend on past frames! */
-  if (pos > 8192) pos -= 8192;
-  else pos = 0;
-#endif
-
-  const int64_t timestamp = pos / av_q2d(priv->formatContext->streams[priv->audioStream]->time_base) / priv->samplerate;
-  dbg(2, "seek frame:%"PRIi64" - idx:%"PRIi64, pos, timestamp);
-  
-  av_seek_frame(priv->formatContext, priv->audioStream, timestamp, AVSEEK_FLAG_ANY | AVSEEK_FLAG_BACKWARD);
-  avcodec_flush_buffers(priv->codecContext);
-  return pos;
-}
-
-static int ad_eval_ffmpeg(const char *f) { 
-  char *ext = strrchr(f, '.');
-  if (!ext) return 10;
-  // libavformat.. guess_format.. 
-  return 40;
-}
-#endif
-
-static const ad_plugin ad_ffmpeg = {
-#ifdef HAVE_FFMPEG
-  &ad_eval_ffmpeg,
-  &ad_open_ffmpeg,
-  &ad_close_ffmpeg,
-  &ad_info_ffmpeg,
-  &ad_seek_ffmpeg,
-  &ad_read_ffmpeg
-#else
-  &ad_eval_null,
-  &ad_open_null,
-  &ad_close_null,
-  &ad_info_null,
-  &ad_seek_null,
-  &ad_read_null
-#endif
-};
-
-/* dlopen handler */
-const ad_plugin * adp_get_ffmpeg() {
-#ifdef HAVE_FFMPEG
-  static int ffinit = 0;
-  if (!ffinit) {
-    ffinit=1;
-#if LIBAVFORMAT_VERSION_INT < AV_VERSION_INT(53, 5, 0)
-    avcodec_init();
-#endif
-    av_register_all();
-    avcodec_register_all();
-    if(ad_debug_level <= 1)
-      av_log_set_level(AV_LOG_QUIET);
-    else 
-      av_log_set_level(AV_LOG_VERBOSE);
-  }
-#endif
-  return &ad_ffmpeg;
-}

source/modules/audio_decoder/ad_plugin.c

@@ -1,176 +0,0 @@
-/**
-   Copyright (C) 2011-2013 Robin Gareus <robin@gareus.org>
-
-   This program is free software; you can redistribute it and/or modify
-   it under the terms of the GNU Lesser Public License as published by
-   the Free Software Foundation; either version 2.1, or (at your option)
-   any later version.
-
-   This program is distributed in the hope that it will be useful,
-   but WITHOUT ANY WARRANTY; without even the implied warranty of
-   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-   GNU Lesser Public License for more details.
-
-   You should have received a copy of the GNU Lesser General Public
-   License along with this library; if not, write to the Free Software
-   Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-*/
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <stdarg.h>
-#include <string.h>
-#include <unistd.h>
-#include <math.h>
-
-#include "ad_plugin.h"
-
-int ad_debug_level = 0;
-
-#define UNUSED(x) (void)(x)
-
-int     ad_eval_null(const char *f) { UNUSED(f); return -1; }
-void *  ad_open_null(const char *f, struct adinfo *n) { UNUSED(f); UNUSED(n); return NULL; }
-int     ad_close_null(void *x) { UNUSED(x); return -1; }
-int     ad_info_null(void *x, struct adinfo *n) { UNUSED(x); UNUSED(n); return -1; }
-int64_t ad_seek_null(void *x, int64_t p) { UNUSED(x); UNUSED(p); return -1; }
-ssize_t ad_read_null(void *x, float*d, size_t s) { UNUSED(x); UNUSED(d); UNUSED(s); return -1;}
-
-typedef struct {
-	ad_plugin const *b; ///< decoder back-end
-	void *d; ///< backend data
-} adecoder;
-
-/* samplecat api */
-
-void ad_init() { /* global init */ }
-
-static ad_plugin const * choose_backend(const char *fn) {
-	int max, val;
-	ad_plugin const *b=NULL;
-	max=0;
-
-	val=adp_get_sndfile()->eval(fn);
-	if (val>max) {max=val; b=adp_get_sndfile();}
-
-	val=adp_get_ffmpeg()->eval(fn);
-	if (val>max) {max=val; b=adp_get_ffmpeg();}
-
-	return b;
-}
-
-void *ad_open(const char *fn, struct adinfo *nfo) {
-	adecoder *d = (adecoder*) calloc(1, sizeof(adecoder));
-	ad_clear_nfo(nfo);
-
-	d->b = choose_backend(fn);
-	if (!d->b) {
-		dbg(0, "fatal: no decoder backend available");
-		free(d);
-		return NULL;
-	}
-	d->d = d->b->open(fn, nfo);
-	if (!d->d) {
-		free(d);
-		return NULL;
-	}
-	return (void*)d;
-}
-
-int ad_info(void *sf, struct adinfo *nfo) {
-	adecoder *d = (adecoder*) sf;
-	if (!d) return -1;
-	return d->b->info(d->d, nfo);
-}
-
-int ad_close(void *sf) {
-	adecoder *d = (adecoder*) sf;
-	if (!d) return -1;
-	int rv = d->b->close(d->d);
-	free(d);
-	return rv;
-}
-
-int64_t ad_seek(void *sf, int64_t pos) {
-	adecoder *d = (adecoder*) sf;
-	if (!d) return -1;
-	return d->b->seek(d->d, pos);
-}
-
-ssize_t ad_read(void *sf, float* out, size_t len){
-	adecoder *d = (adecoder*) sf;
-	if (!d) return -1;
-	return d->b->read(d->d, out, len);
-}
-
-/*
- *  side-effects: allocates buffer
- */
-ssize_t ad_read_mono_dbl(void *sf, struct adinfo *nfo, double* d, size_t len){
-	unsigned int c,f;
-	unsigned int chn = nfo->channels;
-	if (len<1) return 0;
-
-	static float *buf = NULL;
-	static size_t bufsiz = 0;
-	if (!buf || bufsiz != len*chn) {
-		bufsiz=len*chn;
-		buf = (float*) realloc((void*)buf, bufsiz * sizeof(float));
-	}
-
-	len = ad_read(sf, buf, bufsiz);
-
-	for (f=0;f< (len/chn);f++) {
-		double val=0.0;
-		for (c=0;c<chn;c++) {
-			val+=buf[f*chn + c];
-		}
-		d[f]= val/chn;
-	}
-	return len/chn;
-}
-
-
-int ad_finfo (const char *fn, struct adinfo *nfo) {
-	ad_clear_nfo(nfo);
-	void * sf = ad_open(fn, nfo);
-	return ad_close(sf)?1:0;
-}
-
-void ad_clear_nfo(struct adinfo *nfo) {
-	memset(nfo, 0, sizeof(struct adinfo));
-}
-
-void ad_free_nfo(struct adinfo *nfo) {
-	if (nfo->meta_data) free(nfo->meta_data);
-}
-
-void ad_dump_nfo(int dbglvl, struct adinfo *nfo) {
-	dbg(dbglvl, "sample_rate: %u", nfo->sample_rate);
-	dbg(dbglvl, "channels:    %u", nfo->channels);
-	dbg(dbglvl, "length:      %"PRIi64" ms", nfo->length);
-	dbg(dbglvl, "frames:      %"PRIi64, nfo->frames);
-	dbg(dbglvl, "bit_rate:    %d", nfo->bit_rate);
-	dbg(dbglvl, "bit_depth:   %d", nfo->bit_depth);
-	dbg(dbglvl, "channels:    %u", nfo->channels);
-	dbg(dbglvl, "meta-data:   %s", nfo->meta_data?nfo->meta_data:"-");
-}
-
-void ad_debug_printf(const char* func, int level, const char* format, ...) {
-    va_list args;
-
-    va_start(args, format);
-    if (level <= ad_debug_level) {
-        fprintf(stderr, "%s(): ", func);
-        vfprintf(stderr, format, args);
-        fprintf(stderr, "\n");
-    }
-    va_end(args);
-}
-
-void ad_set_debuglevel(int lvl) {
-	ad_debug_level = lvl;
-	if (ad_debug_level<-1) ad_debug_level=-1;
-	if (ad_debug_level>3) ad_debug_level=3;
-}

source/modules/audio_decoder/ad_plugin.h

@@ -1,64 +0,0 @@
-/**
-   Copyright (C) 2011-2013 Robin Gareus <robin@gareus.org>
-
-   This program is free software; you can redistribute it and/or modify
-   it under the terms of the GNU Lesser Public License as published by
-   the Free Software Foundation; either version 2.1, or (at your option)
-   any later version.
-
-   This program is distributed in the hope that it will be useful,
-   but WITHOUT ANY WARRANTY; without even the implied warranty of
-   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-   GNU Lesser Public License for more details.
-
-   You should have received a copy of the GNU Lesser General Public
-   License along with this library; if not, write to the Free Software
-   Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-*/
-#ifndef __AD_PLUGIN_H__
-#define __AD_PLUGIN_H__
-#include <stdint.h>
-#include "audio_decoder/ad.h"
-
-#define dbg(A, B, ...) ad_debug_printf(__func__, A, B, ##__VA_ARGS__)
-
-#ifndef __PRI64_PREFIX
-#if (defined __X86_64__ || defined __LP64__)
-# define __PRI64_PREFIX  "l"
-#else
-# define __PRI64_PREFIX  "ll"
-#endif
-#endif
-
-#ifndef PRIu64
-# define PRIu64   __PRI64_PREFIX "u"
-#endif
-#ifndef PRIi64
-# define PRIi64   __PRI64_PREFIX "i"
-#endif
-
-extern int ad_debug_level;
-
-void ad_debug_printf(const char* func, int level, const char* format, ...);
-
-typedef struct {
-	int     (*eval)(const char *);
-	void *  (*open)(const char *, struct adinfo *);
-	int     (*close)(void *);
-	int     (*info)(void *, struct adinfo *);
-	int64_t (*seek)(void *, int64_t);
-	ssize_t (*read)(void *, float *, size_t);
-} ad_plugin;
-
-int     ad_eval_null(const char *);
-void *  ad_open_null(const char *, struct adinfo *);
-int     ad_close_null(void *);
-int     ad_info_null(void *, struct adinfo *);
-int64_t ad_seek_null(void *, int64_t);
-ssize_t ad_read_null(void *, float*, size_t);
-
-/* hardcoded backends */
-const ad_plugin * adp_get_sndfile();
-const ad_plugin * adp_get_ffmpeg();
-#endif

source/modules/audio_decoder/ad_soundfile.c

@@ -1,154 +0,0 @@
-/**
-   Copyright (C) 2011-2013 Robin Gareus <robin@gareus.org>
-
-   This program is free software; you can redistribute it and/or modify
-   it under the terms of the GNU Lesser Public License as published by
-   the Free Software Foundation; either version 2.1, or (at your option)
-   any later version.
-
-   This program is distributed in the hope that it will be useful,
-   but WITHOUT ANY WARRANTY; without even the implied warranty of
-   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-   GNU Lesser Public License for more details.
-
-   You should have received a copy of the GNU Lesser General Public
-   License along with this library; if not, write to the Free Software
-   Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-*/
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <strings.h>
-#include <unistd.h>
-#include <math.h>
-#include <sndfile.h>
-
-#include "ad_plugin.h"
-
-/* internal abstraction */
-
-typedef struct {
-	SF_INFO sfinfo;
-	SNDFILE *sffile;
-} sndfile_audio_decoder;
-
-static int parse_bit_depth(int format) {
-	/* see http://www.mega-nerd.com/libsndfile/api.html */
-	switch (format&0x0f) {
-		case SF_FORMAT_PCM_S8: return 8;
-		case SF_FORMAT_PCM_16: return 16; /* Signed 16 bit data */
-		case SF_FORMAT_PCM_24: return 24; /* Signed 24 bit data */
-		case SF_FORMAT_PCM_32: return 32; /* Signed 32 bit data */
-		case SF_FORMAT_PCM_U8: return 8;  /* Unsigned 8 bit data (WAV and RAW only) */
-		case SF_FORMAT_FLOAT : return 32; /* 32 bit float data */
-		case SF_FORMAT_DOUBLE: return 64; /* 64 bit float data */
-		default: break;
-	}
-	return 0;
-}
-
-static int ad_info_sndfile(void *sf, struct adinfo *nfo) {
-	sndfile_audio_decoder *priv = (sndfile_audio_decoder*) sf;
-	if (!priv) return -1;
-	if (nfo) {
-		nfo->channels    = priv->sfinfo.channels;
-		nfo->frames      = priv->sfinfo.frames;
-		nfo->sample_rate = priv->sfinfo.samplerate;
-		nfo->length      = priv->sfinfo.samplerate ? (priv->sfinfo.frames * 1000) / priv->sfinfo.samplerate : 0;
-		nfo->bit_depth   = parse_bit_depth(priv->sfinfo.format);
-		nfo->bit_rate    = nfo->bit_depth * nfo->channels * nfo->sample_rate;
-		nfo->meta_data   = NULL;
-	}
-	return 0;
-}
-
-static void *ad_open_sndfile(const char *fn, struct adinfo *nfo) {
-	sndfile_audio_decoder *priv = (sndfile_audio_decoder*) calloc(1, sizeof(sndfile_audio_decoder));
-	priv->sfinfo.format=0;
-	if(!(priv->sffile = sf_open(fn, SFM_READ, &priv->sfinfo))){
-		dbg(0, "unable to open file '%s'.", fn);
-		puts(sf_strerror(NULL));
-		int e = sf_error(NULL);
-		dbg(0, "error=%i", e);
-		free(priv);
-		return NULL;
-	}
-	ad_info_sndfile(priv, nfo);
-	return (void*) priv;
-}
-
-static int ad_close_sndfile(void *sf) {
-	sndfile_audio_decoder *priv = (sndfile_audio_decoder*) sf;
-	if (!priv) return -1;
-	if(!sf || sf_close(priv->sffile)) { 
-		dbg(0, "fatal: bad file close.\n");
-		return -1;
-	}
-	free(priv);
-	return 0;
-}
-
-static int64_t ad_seek_sndfile(void *sf, int64_t pos) {
-	sndfile_audio_decoder *priv = (sndfile_audio_decoder*) sf;
-	if (!priv) return -1;
-	return sf_seek(priv->sffile, pos, SEEK_SET);
-}
-
-static ssize_t ad_read_sndfile(void *sf, float* d, size_t len) {
-	sndfile_audio_decoder *priv = (sndfile_audio_decoder*) sf;
-	if (!priv) return -1;
-	return sf_read_float (priv->sffile, d, len);
-}
-
-static int ad_eval_sndfile(const char *f) { 
-	char *ext = strrchr(f, '.');
-	if (!ext) return 5;
-	/* see http://www.mega-nerd.com/libsndfile/ */
-	if (!strcasecmp(ext, ".wav")) return 100;
-	if (!strcasecmp(ext, ".aiff")) return 100;
-	if (!strcasecmp(ext, ".aifc")) return 100;
-	if (!strcasecmp(ext, ".snd")) return 100;
-	if (!strcasecmp(ext, ".au")) return 100;
-	if (!strcasecmp(ext, ".paf")) return 100;
-	if (!strcasecmp(ext, ".iff")) return 100;
-	if (!strcasecmp(ext, ".svx")) return 100;
-	if (!strcasecmp(ext, ".sf")) return 100;
-	if (!strcasecmp(ext, ".vcc")) return 100;
-	if (!strcasecmp(ext, ".w64")) return 100;
-	if (!strcasecmp(ext, ".mat4")) return 100;
-	if (!strcasecmp(ext, ".mat5")) return 100;
-	if (!strcasecmp(ext, ".pvf5")) return 100;
-	if (!strcasecmp(ext, ".xi")) return 100;
-	if (!strcasecmp(ext, ".htk")) return 100;
-	if (!strcasecmp(ext, ".pvf")) return 100;
-	if (!strcasecmp(ext, ".sd2")) return 100;
-// libsndfile >= 1.0.18
-	if (!strcasecmp(ext, ".flac")) return 80;
-	if (!strcasecmp(ext, ".ogg")) return 80;
-	return 0;
-}
-
-static const ad_plugin ad_sndfile = {
-#if 1
-  &ad_eval_sndfile,
-	&ad_open_sndfile,
-	&ad_close_sndfile,
-	&ad_info_sndfile,
-	&ad_seek_sndfile,
-	&ad_read_sndfile
-#else
-  &ad_eval_null,
-	&ad_open_null,
-	&ad_close_null,
-	&ad_info_null,
-	&ad_seek_null,
-	&ad_read_null
-#endif
-};
-
-/* dlopen handler */
-const ad_plugin * adp_get_sndfile() {
-	return &ad_sndfile;
-}

source/modules/audio_decoder/ffcompat.h

@@ -1,95 +0,0 @@
-/* ffmpeg compatibility wrappers
- *
- * Copyright 2012,2013 Robin Gareus <robin@gareus.org>
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * 1. Redistributions of source code must retain the above copyright notice, this
- *    list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- *    this list of conditions and the following disclaimer in the documentation
- *    and/or other materials provided with the distribution.
- *
- * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" AND
- * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
- * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
- * DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE LIABLE FOR
- * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
- * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
- * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
- * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
- * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
- * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-#ifndef FFCOMPAT_H
-#define FFCOMPAT_H
-
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavutil/avutil.h>
-
-#ifndef AVCODEC_MAX_AUDIO_FRAME_SIZE
-#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000
-#endif
-
-#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(50, 0, 0)
-#define AVMEDIA_TYPE_AUDIO CODEC_TYPE_AUDIO
-#endif
-
-#if LIBAVFORMAT_VERSION_INT < AV_VERSION_INT(53, 2, 0)
-static inline int avformat_open_input(AVFormatContext **ps, const char *filename, void *fmt, void **options)
-{
-	return av_open_input_file(ps, filename, NULL, 0, NULL);
-}
-#endif /* avformat < 53.2.0 */
-
-#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(53, 5, 0)
-static inline AVCodecContext *
-avcodec_alloc_context3(AVCodec *codec __attribute__((unused)))
-{
-	return avcodec_alloc_context();
-}
-
-static inline AVStream *
-avformat_new_stream(AVFormatContext *s, AVCodec *c) {
-	return av_new_stream(s,0);
-}
-
-static inline int
-avcodec_get_context_defaults3(AVCodecContext *s, AVCodec *codec)
-{
-	avcodec_get_context_defaults(s);
-	return 0;
-}
-
-#endif /* < 53.5.0 */
-
-#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(53, 5, 6)
-static inline int
-avcodec_open2(AVCodecContext *avctx, AVCodec *codec, void **options __attribute__((unused)))
-{
-	return avcodec_open(avctx, codec);
-}
-#endif /* <= 53.5.6 */
-
-#if LIBAVFORMAT_VERSION_INT < AV_VERSION_INT(53, 5, 0)
-static inline int
-avformat_find_stream_info(AVFormatContext *ic, void **options)
-{
-	return av_find_stream_info(ic);
-}
-
-static inline void
-avformat_close_input(AVFormatContext **s)
-{
-	av_close_input_file(*s);
-}
-
-#endif /* < 53.5.0 */
-
-#endif /* FFCOMPAT_H */

source/modules/native-plugins/Makefile

@@ -16,29 +16,11 @@ BUILD_CXX_FLAGS += -I. -I../../includes -I../../utils -I../distrho -isystem ..
 
 ALL_C_FLAGS  = $(BUILD_C_FLAGS)
 
-# AudioFile
-ifeq ($(HAVE_AF_DEPS),true)
-ALL_C_FLAGS += -DWANT_AUDIOFILE
-endif
-
-# MidiFile
-ifeq ($(HAVE_MF_DEPS),true)
-ALL_C_FLAGS += -DWANT_MIDIFILE
-endif
-
 # ZynAddSubFX
 ifeq ($(HAVE_ZYN_DEPS),true)
 ALL_C_FLAGS += -DWANT_ZYNADDSUBFX
 endif
 
-# --------------------------------------------------------------
-# Flags for MidiFile
-
-ifeq ($(HAVE_MF_DEPS),true)
-MF_CXX_FLAGS  = $(BUILD_CXX_FLAGS)
-MF_CXX_FLAGS += $(shell pkg-config --cflags smf)
-endif
-
 # --------------------------------------------------------------
 # Flags for ZynAddSubFX
 
@@ -71,6 +53,10 @@ OBJS += \
 	midi-transpose.c.o \
 	nekofilter.c.o
 
+OBJS += \
+	audio-file.cpp.o \
+	midi-file.cpp.o
+
 ifneq ($(WIN32),true)
 # --------------------------------------------------------------
 # External-UI plugins
@@ -88,20 +74,6 @@ OBJS += \
 	juce-patchbay.cpp.o
 endif
 
-# --------------------------------------------------------------
-# AudioFile
-
-ifeq ($(HAVE_AF_DEPS),true)
-OBJS += audio-file.cpp.o
-endif
-
-# --------------------------------------------------------------
-# MidiFile
-
-ifeq ($(HAVE_MF_DEPS),true)
-OBJS += midi-file.cpp.o
-endif
-
 # --------------------------------------------------------------
 # ZynAddSubFX
 
@@ -195,7 +167,7 @@ juce-patchbay.cpp.o: juce-patchbay.cpp
 	$(CXX) $< $(BUILD_CXX_FLAGS) -c -o $@
 
 midi-file.cpp.o: midi-file.cpp midi-base.hpp $(CXXDEPS)
-	$(CXX) $< $(MF_CXX_FLAGS) -c -o $@
+	$(CXX) $< $(BUILD_CXX_FLAGS) -c -o $@
 
 midi-sequencer.cpp.o: midi-sequencer.cpp midi-base.hpp $(CXXDEPS)
 	$(CXX) $< $(BUILD_CXX_FLAGS) -c -o $@

source/modules/native-plugins/_all.c

@@ -26,6 +26,12 @@ extern void carla_register_native_plugin_midithrough();
 extern void carla_register_native_plugin_miditranspose();
 extern void carla_register_native_plugin_nekofilter();
 
+// Audio File
+extern void carla_register_native_plugin_audiofile();
+
+// MIDI File
+extern void carla_register_native_plugin_midifile();
+
 #ifndef CARLA_OS_WIN
 // Carla
 extern void carla_register_native_plugin_carla();
@@ -42,19 +48,6 @@ extern void carla_register_native_plugin_vex_fx();
 extern void carla_register_native_plugin_vex_synth();
 #endif
 
-#ifdef WANT_AUDIOFILE
-// Audio File
-extern void carla_register_native_plugin_audiofile();
-#endif
-
-#ifdef WANT_MIDIFILE
-// MIDI File
-extern void carla_register_native_plugin_midifile();
-#endif
-
-// SunVox File
-extern void carla_register_native_plugin_sunvoxfile();
-
 #ifdef WANT_ZYNADDSUBFX
 // ZynAddSubFX
 extern void carla_register_native_plugin_zynaddsubfx_fx();
@@ -72,6 +65,12 @@ void carla_register_all_plugins()
     carla_register_native_plugin_miditranspose();
     carla_register_native_plugin_nekofilter();
 
+    // Audio File
+    carla_register_native_plugin_audiofile();
+
+    // MIDI File
+    carla_register_native_plugin_midifile();
+
 #ifndef CARLA_OS_WIN
     // Carla
     carla_register_native_plugin_carla();
@@ -88,16 +87,6 @@ void carla_register_all_plugins()
     carla_register_native_plugin_vex_synth();
 #endif
 
-#ifdef WANT_AUDIOFILE
-    // Audio File
-    carla_register_native_plugin_audiofile();
-#endif
-
-#ifdef WANT_MIDIFILE
-    // MIDI File
-    carla_register_native_plugin_midifile();
-#endif
-
 #ifdef WANT_ZYNADDSUBFX
     // ZynAddSubFX
     carla_register_native_plugin_zynaddsubfx_fx();

source/plugin/Makefile

@@ -69,10 +69,6 @@ LIBS += ../modules/juce_audio_basics.a
 LIBS += ../modules/juce_core.a
 LIBS += ../modules/rtmempool.a
 
-ifeq ($(HAVE_AF_DEPS),true)
-LIBS += ../modules/audio_decoder.a
-endif
-
 ifeq ($(HAVE_JUCE_UI),true)
 LIBS += ../modules/juce_audio_processors.a
 LIBS += ../modules/juce_data_structures.a