falkTX
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Carla
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remove audio_decoder, cleanup

tags/1.9.4
falkTX 11 years ago
parent
e1e3f9fe2d
commit
5dc301b5d6
18 changed files with 23 additions and 1493 deletions
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  1. +0
    -31
      source/Makefile.mk
  2. +3
    -17
      source/backend/CarlaStandalone.cpp
  3. +0
    -4
      source/backend/Makefile
  4. +2
    -21
      source/backend/engine/CarlaEngine.cpp
  5. +0
    -15
      source/backend/plugin/NativePlugin.cpp
  6. +0
    -13
      source/bridges/Makefile
  7. +0
    -286
      source/includes/sunvox/sunvox.h
  8. +1
    -4
      source/modules/Makefile
  9. +0
    -53
      source/modules/audio_decoder/Makefile
  10. +0
    -127
      source/modules/audio_decoder/ad.h
  11. +0
    -373
      source/modules/audio_decoder/ad_ffmpeg.c
  12. +0
    -176
      source/modules/audio_decoder/ad_plugin.c
  13. +0
    -64
      source/modules/audio_decoder/ad_plugin.h
  14. +0
    -154
      source/modules/audio_decoder/ad_soundfile.c
  15. +0
    -95
      source/modules/audio_decoder/ffcompat.h
  16. +5
    -33
      source/modules/native-plugins/Makefile
  17. +12
    -23
      source/modules/native-plugins/_all.c
  18. +0
    -4
      source/plugin/Makefile

+ 0
- 31
source/Makefile.mk View File

@@ -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)


+ 3
- 17
source/backend/CarlaStandalone.cpp View File

@@ -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


+ 0
- 4
source/backend/Makefile View File

@@ -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


+ 2
- 21
source/backend/engine/CarlaEngine.cpp View File

@@ -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
}

// -------------------------------------------------------------------


+ 0
- 15
source/backend/plugin/NativePlugin.cpp View File

@@ -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


+ 0
- 13
source/bridges/Makefile View File

@@ -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


+ 0
- 286
source/includes/sunvox/sunvox.h View File

@@ -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

+ 1
- 4
source/modules/Makefile View File

@@ -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

# --------------------------------------------------------------

+ 0
- 53
source/modules/audio_decoder/Makefile View File

@@ -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 $@

# --------------------------------------------------------------

+ 0
- 127
source/modules/audio_decoder/ad.h View File

@@ -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

+ 0
- 373
source/modules/audio_decoder/ad_ffmpeg.c View File

@@ -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;
}

+ 0
- 176
source/modules/audio_decoder/ad_plugin.c View File

@@ -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;
}

+ 0
- 64
source/modules/audio_decoder/ad_plugin.h View File

@@ -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

+ 0
- 154
source/modules/audio_decoder/ad_soundfile.c View File

@@ -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;
}

+ 0
- 95
source/modules/audio_decoder/ffcompat.h View File

@@ -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 */

+ 5
- 33
source/modules/native-plugins/Makefile View File

@@ -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 $@


+ 12
- 23
source/modules/native-plugins/_all.c View File

@@ -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();


+ 0
- 4
source/plugin/Makefile View File

@@ -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


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