Changeset 111
- Timestamp:
- 09/03/06 21:19:57 (5 years ago)
- Files:
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- 1 added
- 2 removed
- 19 modified
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fm_gui/trunk/AUTHORS (modified) (1 diff)
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fm_gui/trunk/README (modified) (1 diff)
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fm_gui/trunk/data/preferences.glade (added)
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fm_gui/trunk/src/Makefile.am (modified) (1 diff)
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fm_gui/trunk/src/Makefile.in (modified) (1 diff)
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fm_gui/trunk/src/base.c (modified) (4 diffs)
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fm_gui/trunk/src/main.c (modified) (3 diffs)
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fm_server/trunk/AUTHORS (modified) (1 diff)
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fm_server/trunk/libtool (modified) (6 diffs)
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fm_server/trunk/src/Makefile.am (modified) (2 diffs)
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fm_server/trunk/src/Makefile.in (modified) (5 diffs)
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fm_server/trunk/src/fm_effect_rate.c (modified) (1 diff)
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fm_server/trunk/src/fm_output_alsa.c (modified) (14 diffs)
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fm_server/trunk/src/fm_output_alsa.h (modified) (3 diffs)
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fm_server/trunk/src/fm_output_oss.c (deleted)
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fm_server/trunk/src/fm_output_oss.h (deleted)
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fm_server/trunk/src/main.c (modified) (2 diffs)
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fm_server/trunk/test/test_play_file.c (modified) (2 diffs)
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fm_server/trunk/test/test_simple.c (modified) (2 diffs)
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rgc/trunk/AUTHORS (modified) (2 diffs)
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rgc/trunk/intltool-merge (modified) (1 diff)
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rgc/trunk/intltool-update (modified) (1 diff)
Legend:
- Unmodified
- Added
- Removed
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fm_gui/trunk/AUTHORS
r44 r111 1 Main author and developer: 2 Alberto Botti <ath9@libero.it> 3 4 Based on code and ideas from the GDAM "Geoff and Dave's Audio Mixer" 5 (www.ffem.org/gdam), the GStreamer multimedia framework (www.gstreamer.net) 6 and the terminatorX realtime audio scrathing program by Alexander König 7 (www.terminatorX.cx). 1 Main author and mantainer: 2 Alberto Botti <alberto.botti@yoda2000.net> -
fm_gui/trunk/README
r44 r111 1 fm_gui 2 3 a graphical interface for the fm_mixer, part of the FreeMix project 4 http://ciampix.net:8080/proj/wiki/FreeMix -
fm_gui/trunk/src/Makefile.am
r103 r111 12 12 13 13 14 ##AM_CFLAGS = $(BINRELOC_CFLAGS)14 AM_CFLAGS = $(BINRELOC_CFLAGS) 15 15 16 16 AM_LDFLAGS = $(FM_GUI_LIBS) -
fm_gui/trunk/src/Makefile.in
r103 r111 226 226 SUBDIRS = tx_widgets 227 227 INCLUDES = $(FM_GUI_CFLAGS) 228 AM_CFLAGS = $(BINRELOC_CFLAGS) 228 229 AM_LDFLAGS = $(FM_GUI_LIBS) 229 230 fm_gui_SOURCES = \ -
fm_gui/trunk/src/base.c
r87 r111 9 9 #include "fm_djmixer_gui.h" 10 10 #include "fm_djmixer.h" 11 12 13 extern gchar *global_backend; 14 extern gchar *BACKEND_ALSA; 15 extern gchar *BACKEND_JACK; 11 16 12 17 … … 48 53 g_usleep(100); 49 54 rgc_remote_object_call (bdata->pipeline, "run", NULL); 50 g_usleep(100); 51 rgc_remote_object_call (obj, "connect_to_hw_ports", NULL); 55 if (global_backend == BACKEND_JACK) { 56 g_usleep(100); 57 rgc_remote_object_call (obj, "connect_to_hw_ports", NULL); 58 } 52 59 53 60 /* check if a mixer already exists */ … … 67 74 /* check if an output already exists */ 68 75 /* if not, build one */ 69 rgc_remote_object_new_async (base_output_created_cb, data, 70 "FmOutputJack", "output", NULL); 76 77 if (global_backend == BACKEND_JACK) 78 rgc_remote_object_new_async (base_output_created_cb, data, 79 "FmOutputJack", "output", NULL); 80 else if (global_backend == BACKEND_ALSA) 81 rgc_remote_object_new_async (base_output_created_cb, data, 82 "FmOutputAlsa", "output", NULL); 83 else 84 g_assert_not_reached (); 71 85 } 72 86 … … 79 93 rgc_remote_object_new_async (base_pipeline_created_cb, bdata, 80 94 "FmPipeline", "pipeline", NULL); 95 96 return TRUE; 81 97 } -
fm_gui/trunk/src/main.c
r103 r111 8 8 9 9 10 gchar *global_backend; 11 gchar *BACKEND_ALSA = "ALSA"; 12 gchar *BACKEND_JACK = "JACK"; 10 13 11 14 … … 28 31 { NULL } 29 32 }; 33 34 // global_backend = BACKEND_JACK; 35 global_backend = BACKEND_ALSA; 30 36 31 37 GOptionContext *context = g_option_context_new ("- a GUI for the fm_server"); … … 35 41 36 42 global_init (); 37 38 43 39 44 /* Initialize a connection. We use RBP (on a TCP socket) here. */ -
fm_server/trunk/AUTHORS
r10 r111 1 1 Main author and developer: 2 Alberto Botti <a th9@libero.it>2 Alberto Botti <alberto.botti@yoda2000.net> 3 3 4 Based on code and ideas from the GDAM "Geoff and Dave's Audio Mixer" 5 (www.ffem.org/gdam), the GStreamer multimedia framework (www.gstreamer.net) 6 and the terminatorX realtime audio scrathing program by Alexander König 7 (www.terminatorX.cx). 4 Based on code and ideas from: 5 * the GDAM "Geoff and Dave's Audio Mixer" (www.ffem.org/gdam) 6 * the GStreamer multimedia framework (www.gstreamer.net) 7 * the terminatorX realtime audio scrathing program by Alexander König 8 (www.terminatorX.cx) 9 * the JACK Audio Connection Kit (www.jackaudio.com) examples and ringbuffer code 10 * the ALSA (www.alsa-project.org) programming guide -
fm_server/trunk/libtool
r80 r111 45 45 # ### BEGIN LIBTOOL CONFIG 46 46 47 # Libtool was configured on host sarge:47 # Libtool was configured on host crash: 48 48 49 49 # Shell to use when invoking shell scripts. … … 86 86 87 87 # LTCC compiler flags. 88 LTCFLAGS="- g -O2"88 LTCFLAGS="-O0 -g -Wall -Werror" 89 89 90 90 # A language-specific compiler. … … 329 329 330 330 # Run-time system search path for libraries 331 sys_lib_dlsearch_path_spec="/lib /usr/lib "331 sys_lib_dlsearch_path_spec="/lib /usr/lib /usr/X11R6/lib /usr/lib/mozilla-firefox /opt/e/lib " 332 332 333 333 # Fix the shell variable $srcfile for the compiler. … … 7224 7224 # ### BEGIN LIBTOOL TAG CONFIG: CXX 7225 7225 7226 # Libtool was configured on host sarge:7226 # Libtool was configured on host crash: 7227 7227 7228 7228 # Shell to use when invoking shell scripts. … … 7265 7265 7266 7266 # LTCC compiler flags. 7267 LTCFLAGS="- g -O2"7267 LTCFLAGS="-O0 -g -Wall -Werror" 7268 7268 7269 7269 # A language-specific compiler. … … 7505 7505 7506 7506 # Run-time system search path for libraries 7507 sys_lib_dlsearch_path_spec="/lib /usr/lib "7507 sys_lib_dlsearch_path_spec="/lib /usr/lib /usr/X11R6/lib /usr/lib/mozilla-firefox /opt/e/lib " 7508 7508 7509 7509 # Fix the shell variable $srcfile for the compiler. -
fm_server/trunk/src/Makefile.am
r95 r111 44 44 fm_effect_pan.c\ 45 45 fm_effect_pan.h 46 ## fm_output_oss.c\47 ## fm_output_oss.h\48 46 49 47 … … 53 51 54 52 55 ##if HAVE_ALSA 56 ##fm_server_SOURCES += fm_output_alsa.c fm_output_alsa.h 57 ##fm_server_LDADD += -lasound 58 ##endif 53 if HAVE_ALSA 54 fm_server_SOURCES += \ 55 fm_output_alsa.c \ 56 fm_output_alsa.h 57 58 fm_server_LDADD += -lasound 59 endif 59 60 60 61 -
fm_server/trunk/src/Makefile.in
r95 r111 38 38 host_triplet = @host@ 39 39 bin_PROGRAMS = fm_server$(EXEEXT) 40 @HAVE_JACK_TRUE@am__append_1 = \ 40 @HAVE_ALSA_TRUE@am__append_1 = \ 41 @HAVE_ALSA_TRUE@ fm_output_alsa.c \ 42 @HAVE_ALSA_TRUE@ fm_output_alsa.h 43 44 @HAVE_ALSA_TRUE@am__append_2 = -lasound 45 @HAVE_JACK_TRUE@am__append_3 = \ 41 46 @HAVE_JACK_TRUE@ fm_output_jack.c \ 42 47 @HAVE_JACK_TRUE@ fm_output_jack.h … … 62 67 fm_effect_gain.h fm_effect_rate.c fm_effect_rate.h fm_mixer.c \ 63 68 fm_mixer.h fm_djmixer.c fm_djmixer.h fm_buffer.c fm_buffer.h \ 64 fm_effect_pan.c fm_effect_pan.h fm_output_jack.c \ 65 fm_output_jack.h 66 @HAVE_JACK_TRUE@am__objects_1 = fm_output_jack.$(OBJEXT) 69 fm_effect_pan.c fm_effect_pan.h fm_output_alsa.c \ 70 fm_output_alsa.h fm_output_jack.c fm_output_jack.h 71 @HAVE_ALSA_TRUE@am__objects_1 = fm_output_alsa.$(OBJEXT) 72 @HAVE_JACK_TRUE@am__objects_2 = fm_output_jack.$(OBJEXT) 67 73 am_fm_server_OBJECTS = main.$(OBJEXT) fm_utils.$(OBJEXT) \ 68 74 fm_input.$(OBJEXT) fm_output.$(OBJEXT) fm_effect.$(OBJEXT) \ … … 72 78 fm_effect_rate.$(OBJEXT) fm_mixer.$(OBJEXT) \ 73 79 fm_djmixer.$(OBJEXT) fm_buffer.$(OBJEXT) \ 74 fm_effect_pan.$(OBJEXT) $(am__objects_1) 80 fm_effect_pan.$(OBJEXT) $(am__objects_1) $(am__objects_2) 75 81 fm_server_OBJECTS = $(am_fm_server_OBJECTS) 76 82 am__DEPENDENCIES_1 = 77 83 fm_server_DEPENDENCIES = $(am__DEPENDENCIES_1) \ 78 fm_input_gst/libfm_input_gst.la 84 fm_input_gst/libfm_input_gst.la $(am__DEPENDENCIES_1) 79 85 DEFAULT_INCLUDES = -I. -I$(srcdir) -I$(top_builddir) 80 86 depcomp = $(SHELL) $(top_srcdir)/depcomp … … 258 264 fm_effect_rate.c fm_effect_rate.h fm_mixer.c fm_mixer.h \ 259 265 fm_djmixer.c fm_djmixer.h fm_buffer.c fm_buffer.h \ 260 fm_effect_pan.c fm_effect_pan.h $(am__append_1) 261 fm_server_LDADD = \ 262 $(FM_SERVER_LIBS) \ 263 fm_input_gst/libfm_input_gst.la 264 266 fm_effect_pan.c fm_effect_pan.h $(am__append_1) \ 267 $(am__append_3) 268 fm_server_LDADD = $(FM_SERVER_LIBS) fm_input_gst/libfm_input_gst.la \ 269 $(am__append_2) 265 270 all: all-recursive 266 271 … … 347 352 @AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/fm_mixer.Po@am__quote@ 348 353 @AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/fm_output.Po@am__quote@ 354 @AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/fm_output_alsa.Po@am__quote@ 349 355 @AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/fm_output_jack.Po@am__quote@ 350 356 @AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/fm_pipeline.Po@am__quote@ -
fm_server/trunk/src/fm_effect_rate.c
r109 r111 140 140 fm_effect_rate_init (FmEffectRate *effect_rate) 141 141 { 142 effect_rate->converter = 1;142 effect_rate->converter = 3; 143 143 effect_rate->change_rate = -1; 144 144 effect_rate->rate = 1.00; -
fm_server/trunk/src/fm_output_alsa.c
r10 r111 3 3 #include <glib.h> 4 4 #include <glib-object.h> 5 #include <rgc.h> 5 #include <rgc_server.h> 6 6 7 #include <alsa/asoundlib.h> 7 8 … … 12 13 13 14 14 #define ALSA_DEFAULT_DEVICE " hw:0,0"15 #define ALSA_DEFAULT_DEVICE "default" 15 16 #define ALSA_DEFAULT_BUFFER_TIME 22000 16 #define ALSA_DEFAULT_PERIOD_TIME 500017 #define ALSA_DEFAULT_PERIOD_TIME 10000 17 18 18 19 extern gint temp_fd; … … 38 39 static gboolean fm_output_alsa_deactivate (FmElement *element); 39 40 static void fm_output_alsa_finalize (GObject *object); 40 41 static void fm_output_alsa_get_buffer (FmElement *element, 42 FmBuffer *buffer); 43 44 gint xrun_recovery (snd_pcm_t *handle, 41 static gint xrun_recovery (snd_pcm_t *handle, 45 42 gint err); 43 static gpointer fm_output_alsa_play (gpointer data); 44 static gboolean fm_output_alsa_start (FmOutput *output); 45 46 46 47 47 static FmOutputClass *parent_class = NULL; … … 76 76 fm_output_alsa_class_register (void) 77 77 { 78 rgc_class_register ("FmOutputAlsa", 79 FM_TYPE_OUTPUT_ALSA); 80 81 rgc_method_register ("connect_src", 82 FM_TYPE_OUTPUT_ALSA, 83 G_CALLBACK (fm_element_connect_src), 84 rgc_marshal_VOID__ELEMENT, 85 1, G_TYPE_STRING); 78 rgc_class_register (FM_TYPE_OUTPUT_ALSA); 86 79 } 87 80 … … 98 91 gobject_class->set_property = fm_output_alsa_set_property; 99 92 gobject_class->get_property = fm_output_alsa_get_property; 93 100 94 gobject_class->finalize = fm_output_alsa_finalize; 95 // gobject_class->dispose = 101 96 102 97 element_class->activate = fm_output_alsa_activate; 103 98 element_class->deactivate = fm_output_alsa_deactivate; 104 element_class->get_buffer = fm_output_alsa_get_buffer; 105 106 output_class->play = fm_output_alsa_play; 99 100 output_class->start = fm_output_alsa_start; 101 //output_class->stop = fm_output_alsa_stop; 102 107 103 108 104 g_object_class_install_property (gobject_class, … … 145 141 output_alsa->buffer_size = 0; 146 142 output_alsa->period_size = 0; 147 output_alsa->engine_buffer = NULL; 143 output_alsa->thread = NULL; 144 output_alsa->stop_processing = FALSE; 148 145 } 149 146 … … 214 211 FmOutputAlsa *output_alsa = FM_OUTPUT_ALSA (element); 215 212 gint err; 216 217 g_assert ((FmElement *) g_list_nth_data (element->src_list, 0));218 fm_element_activate ((FmElement *) g_list_nth_data (element->src_list, 0));219 213 220 214 g_print ("Activating FmOutputAlsa\n"); 221 215 222 223 216 // Open the device 224 217 … … 265 258 } 266 259 260 output->sample_rate = 44100; 267 261 err = snd_pcm_hw_params_set_rate_near (output_alsa->sound_handle, 268 262 output_alsa->hwparams, … … 272 266 return FALSE; 273 267 } 274 268 269 g_print ("ALSA sample rate: %d\n", output->sample_rate); 270 271 guint n_channels = FM_CHANNELS_STEREO; 275 272 err = snd_pcm_hw_params_set_channels (output_alsa->sound_handle, 276 273 output_alsa->hwparams, 277 output->channels_n);278 if (err < 0) { 279 printf("snd_pcm_hw_params_set_channels %d: %s\n", output->channels_n, snd_strerror (err));280 return FALSE; 281 } 282 283 g uint dir = 0; // ???274 n_channels); 275 if (err < 0) { 276 printf("snd_pcm_hw_params_set_channels %d: %s\n", n_channels, snd_strerror (err)); 277 return FALSE; 278 } 279 280 gint dir = 0; // ??? 284 281 285 282 err = snd_pcm_hw_params_set_buffer_time_near (output_alsa->sound_handle, … … 319 316 } 320 317 321 output->samples_per_buffer = output_alsa->period_size;322 323 318 err = snd_pcm_hw_params (output_alsa->sound_handle, 324 319 output_alsa->hwparams); … … 329 324 } 330 325 331 output->samples_per_buffer = output_alsa->period_size; 332 333 // Our integer buffer 334 335 output_alsa->engine_buffer = fm_buffer_new (MAX_SAMPLES, 336 FM_SAMPLES_INT16, FM_INTERLEAVING_NORMAL, FM_CHANNELS_STEREO); 326 output_alsa->buffer_size_samples = output_alsa->period_size; 327 output_alsa->main_buffer = fm_buffer_new (output_alsa->buffer_size_samples, FM_CHANNELS_STEREO); 328 output_alsa->buffer_int = g_new (gint16, output_alsa->buffer_size_samples * FM_CHANNELS_STEREO); 329 330 /* We postpone this to set the pipeline's sample rate */ 331 if (!fm_element_activate_connected (element)) 332 return FALSE; 337 333 338 334 return TRUE; … … 343 339 fm_output_alsa_deactivate (FmElement *element) 344 340 { 345 FmOutputAlsa *output_alsa = FM_OUTPUT_ALSA (element);341 /*FmOutputAlsa *output_alsa = FM_OUTPUT_ALSA (element); 346 342 gint err; 347 343 … … 362 358 output_alsa->buffer_size = 0; 363 359 output_alsa->period_size = 0; 364 output_alsa->engine_buffer = NULL; 360 output_alsa->engine_buffer = NULL;*/ 365 361 366 362 return TRUE; … … 379 375 380 376 381 382 383 static void 384 fm_output_alsa_get_buffer (FmElement *element, 385 FmBuffer *buffer) 386 { 387 g_return_if_fail (FM_IS_OUTPUT_ALSA (element)); 388 389 g_assert (g_list_nth_data (element->src_list, 0)); 390 /* FIXME */ 391 fm_element_get_buffer (FM_ELEMENT (g_list_nth_data (element->src_list, 0)), 392 buffer); 393 394 return; 395 } 396 397 398 void 399 fm_output_alsa_play (FmOutput *output, 400 FmBuffer *buffer) 401 { 402 FmOutputAlsa *output_alsa = FM_OUTPUT_ALSA (output); 403 404 fm_buffer_convert (buffer, output_alsa->engine_buffer); 405 406 fm_buffer_dump_to_file (output_alsa->engine_buffer, temp_fd); 407 408 gint err = 0; 409 410 g_assert (output_alsa->sound_handle); 411 412 err = snd_pcm_writei (output_alsa->sound_handle, 413 FM_BUFFER_DATA (output_alsa->engine_buffer), 414 FM_BUFFER_USED_SAMPLES (output_alsa->engine_buffer)); 415 if (err < 0) { 416 if (xrun_recovery (output_alsa->sound_handle, err) < 0) { 417 g_print ("alsa: xrun1\n"); 418 return; // FALSE 377 static gboolean 378 fm_output_alsa_start (FmOutput *output) 379 { 380 FmOutputAlsa *output_alsa = FM_OUTPUT_ALSA(output); 381 GError *error = NULL; 382 383 output_alsa->thread = g_thread_create (fm_output_alsa_play, output_alsa, 384 FALSE, &error); 385 if (output_alsa->thread) { 386 g_print("ALSA output started!\n"); 387 return TRUE; 388 } else { 389 g_warning("Failed starting ALSA output, %s", error->message); 390 return FALSE; 391 } 392 } 393 394 395 #define FLOAT_TO_INT(x) ((gint16) ((x) * 32768.0)) 396 397 static gpointer 398 fm_output_alsa_play (gpointer data) 399 { 400 FmElement *element = FM_ELEMENT (data); 401 FmOutputAlsa *output_alsa = FM_OUTPUT_ALSA (data); 402 403 while (!output_alsa->stop_processing) { 404 405 //g_print("asking for %d samples\n", output_alsa->buffer_size_samples); 406 FM_BUFFER_REQUESTED(output_alsa->main_buffer) = output_alsa->buffer_size_samples; 407 fm_element_get_buffer_from_src(element, output_alsa->main_buffer); 408 409 guint i; 410 gint16 *ptr = output_alsa->buffer_int; 411 for (i = 0; i < output_alsa->buffer_size_samples; i++) { 412 *(ptr++) = FLOAT_TO_INT (output_alsa->main_buffer->channel_data[0][i]); 413 *(ptr++) = FLOAT_TO_INT (output_alsa->main_buffer->channel_data[1][i]); 419 414 } 415 416 gint err = 0; 420 417 421 418 err = snd_pcm_writei (output_alsa->sound_handle, 422 FM_BUFFER_DATA (output_alsa->engine_buffer),423 FM_BUFFER_USED_SAMPLES (output_alsa->engine_buffer));424 419 output_alsa->buffer_int, 420 output_alsa->buffer_size_samples); 421 425 422 if (err < 0) { 426 423 if (xrun_recovery (output_alsa->sound_handle, err) < 0) { 427 g_print ("alsa: xrun2\n"); 428 return; // FALSE 424 g_print ("alsa: xrun1\n"); 425 continue; 426 } 427 428 err = snd_pcm_writei (output_alsa->sound_handle, 429 output_alsa->buffer_int, 430 output_alsa->buffer_size_samples); 431 432 if (err < 0) { 433 if (xrun_recovery (output_alsa->sound_handle, err) < 0) { 434 g_print ("alsa: xrun2\n"); 435 continue; 436 } 429 437 } 430 } 431 } 432 return; // TRUE 438 } 439 } 440 441 g_print ("output_alsa->stop_processing == TRUE!\n"); 442 443 return NULL; 433 444 } 434 445 435 446 436 447 // Underrun and suspend recovery 437 gint448 static gint 438 449 xrun_recovery (snd_pcm_t *handle, 439 450 gint err) -
fm_server/trunk/src/fm_output_alsa.h
r10 r111 17 17 gchar *device_name; 18 18 19 GThread *thread; 20 gboolean stop_processing; 21 19 22 snd_pcm_t *sound_handle; 20 23 snd_output_t *errlog; … … 29 32 snd_pcm_uframes_t period_size; 30 33 31 FmBuffer *engine_buffer; 34 guint buffer_size_samples; 35 FmBuffer *main_buffer; 36 gint16 *buffer_int; 32 37 }; 33 38 … … 48 53 fm_output_alsa_new (void); 49 54 50 /*void51 fm_output_alsa_get_buffer (FmElement *element,52 FmBuffer *buffer);*/53 54 void55 fm_output_alsa_play (FmOutput *output,56 FmBuffer *buffer); -
fm_server/trunk/src/main.c
r109 r111 59 59 #include "fm_output_alsa.h" 60 60 #endif 61 #include "fm_output_oss.h"62 61 #include "fm_output_jack.h" 63 62 #include "fm_mixer.h" … … 157 156 fm_mixer_dj_class_register ();*/ 158 157 #if HAVE_ALSA_ASOUNDLIB_H 159 //fm_output_alsa_class_register ();158 fm_output_alsa_class_register (); 160 159 #endif 161 160 //fm_output_oss_class_register (); -
fm_server/trunk/test/test_play_file.c
r29 r111 40 40 rgc_remote_object_call (pipeline, "run", NULL); 41 41 g_usleep(100); 42 rgc_remote_object_call (output, "connect_to_hw_ports", NULL);42 // rgc_remote_object_call (output, "connect_to_hw_ports", NULL); 43 43 g_usleep(100); 44 44 //rgc_remote_object_call (input, "play", NULL); … … 70 70 GValue *device_name = g_value_init_string ("default"); 71 71 rgc_remote_object_new_async (output_created_cb, NULL, 72 "FmOutputJack", "output", NULL); 72 // "FmOutputJack", "output", NULL); 73 "FmOutputAlsa", "output", NULL); 73 74 g_value_destroy (device_name); 74 75 } -
fm_server/trunk/test/test_simple.c
r90 r111 34 34 rgc_remote_object_call (pipeline, "activate", NULL); 35 35 rgc_remote_object_call (pipeline, "run", NULL); 36 rgc_remote_object_call (output, "connect_to_hw_ports", NULL);36 // rgc_remote_object_call (output, "connect_to_hw_ports", NULL); 37 37 38 38 sleep (200); … … 92 92 GValue *device_name = g_value_init_string ("default"); 93 93 rgc_remote_object_new_async (output_created_cb, NULL, 94 "FmOutputJack", "output", NULL); 94 // "FmOutputJack", "output", NULL); 95 "FmOutputAlsa", "output", NULL); 95 96 g_value_destroy (device_name); 96 97 } -
rgc/trunk/AUTHORS
r3 r111 1 1 Main author and mantainer: 2 Alberto Botti <a thproject@hotpop.com>2 Alberto Botti <alberto.botti@yoda2000.net> 3 3 4 4 Based on many ideas and source code from the GLib library: … … 11 11 12 12 Distributed under the terms of the GNU Library General Public License (LGPL) 13 version 2 .13 version 2 or later. -
rgc/trunk/intltool-merge
r100 r111 258 258 { 259 259 my ($encoding) = @_; 260 my $alias_file = $ENV{"G_CHARSET_ALIAS"} || "/ usr/local/lib/charset.alias";260 my $alias_file = $ENV{"G_CHARSET_ALIAS"} || "/home/ath/test/lib/charset.alias"; 261 261 262 262 # seek character encoding aliases in charset.alias (glib) -
rgc/trunk/intltool-update
r100 r111 558 558 sub GenerateHeaders 559 559 { 560 my $EXTRACT = "/home/ath/ Projects/fm/freemix/rgc/trunk/intltool-extract";560 my $EXTRACT = "/home/ath/Desktop/Progetti/fm/freemix/rgc/trunk/intltool-extract"; 561 561 chomp $EXTRACT; 562 562
