diff options
| -rw-r--r-- | ChangeLog | 20 | ||||
| -rw-r--r-- | configure.ac | 33 | ||||
| -rw-r--r-- | ext/Makefile.am | 16 | ||||
| -rw-r--r-- | ext/audioresample/Makefile.am | 10 | ||||
| -rw-r--r-- | ext/audioresample/gstaudioresample.c | 435 | ||||
| -rw-r--r-- | ext/audioresample/gstaudioresample.h | 74 | ||||
| -rw-r--r-- | gst-libs/gst/media-info/media-info-priv.h | 1 |
7 files changed, 574 insertions, 15 deletions
@@ -1,3 +1,23 @@ +2004-12-15 David Schleef <ds@schleef.org> + + * configure.ac: add audioresample and cairo plugins. Remove + HAVE_MMX stuff, because it's not used. + * ext/Makefile.am: same + * ext/audioresample/Makefile.am: You are not ready for an + audio resampling element based on audioresample. + * ext/audioresample/gstaudioresample.c: + * ext/audioresample/gstaudioresample.h: + * ext/cairo/Makefile.am: You are not ready for overlay elements + based on cairo. Don't look too closely, these elements kinda + suck right now. + * ext/cairo/gstcairo.c: new + * ext/cairo/gsttextoverlay.c: new + * ext/cairo/gsttextoverlay.h: new + * ext/cairo/gsttimeoverlay.c: new + * ext/cairo/gsttimeoverlay.h: new + * gst-libs/gst/media-info/media-info-priv.h: fix compile + problem with compilers that don't support variadic macros. + 2004-12-15 Balamurali Viswanathan <balamurali.viswanathan@wipro.com> Reviewed by: David Schleef <ds@schleef.org> diff --git a/configure.ac b/configure.ac index 72514037..e64bea79 100644 --- a/configure.ac +++ b/configure.ac @@ -806,6 +806,22 @@ GST_CHECK_FEATURE(AUDIOFILE, [audiofile], afsink afsrc, [ AC_CHECK_LIB(audiofile, af_virtual_file_new, , HAVE_AUDIOFILE="no") fi]) +dnl *** audioresample *** +translit(dnm, m, l) AM_CONDITIONAL(USE_AUDIORESAMPLE, true) +GST_CHECK_FEATURE(AUDIORESAMPLE, [audioresample plug-in], audioresample, [ + PKG_CHECK_MODULES(AUDIORESAMPLE, audioresample-0.1, HAVE_AUDIORESAMPLE=yes, HAVE_AUDIORESAMPLE=no) + AC_SUBST(AUDIORESAMPLE_CFLAGS) + AC_SUBST(AUDIORESAMPLE_LIBS) +]) + +dnl *** cairo *** +translit(dnm, m, l) AM_CONDITIONAL(USE_CAIRO, true) +GST_CHECK_FEATURE(CAIRO, [cairo plug-in], cairo, [ + PKG_CHECK_MODULES(CAIRO, cairo, HAVE_CAIRO=yes, HAVE_CAIRO=no) + AC_SUBST(CAIRO_CFLAGS) + AC_SUBST(CAIRO_LIBS) +]) + dnl *** cdaudio *** translit(dnm, m, l) AM_CONDITIONAL(USE_CDAUDIO, true) GST_CHECK_FEATURE(CDAUDIO, [cdaudio], cdaudio, [ @@ -1722,15 +1738,6 @@ dnl ###################################################################### dnl # Check command line parameters, and set shell variables accordingly # dnl ###################################################################### -AC_ARG_ENABLE(libmmx, - AC_HELP_STRING([--enable-libmmx],[use libmmx, if available]), -[case "${enableval}" in - yes) USE_LIBMMX=$HAVE_LIBMMX ;; - no) USE_LIBMMX=no ;; - *) AC_MSG_ERROR(bad value ${enableval} for --enable-libmmx) ;; -esac], -[USE_LIBMMX=$HAVE_LIBMMX]) dnl Default value - AC_ARG_ENABLE(atomic, AC_HELP_STRING([--enable-atomic],[use atomic reference counting header]), [case "${enableval}" in @@ -1783,10 +1790,6 @@ dnl # Set defines according to variables set above # dnl ################################################ -if test "x$USE_LIBMMX" = xyes; then - AC_DEFINE(HAVE_LIBMMX, 1, [Define if libmmx is available]) -fi - if test "x$USE_ATOMIC_H" = xyes; then AC_DEFINE(HAVE_ATOMIC_H, 1, [Define if atomic.h header file is available]) fi @@ -1804,8 +1807,6 @@ dnl ############################# dnl These should be "USE_*" instead of "HAVE_*", but some packages expect dnl HAVE_ and it is likely to be easier to stick with the old name -AM_CONDITIONAL(HAVE_LIBMMX, test "x$USE_LIBMMX" = "xyes") - AM_CONDITIONAL(HAVE_ATOMIC_H, test "x$USE_ATOMIC_H" = "xyes") AM_CONDITIONAL(EXPERIMENTAL, test "$EXPERIMENTAL" = "$xyes") @@ -1980,6 +1981,8 @@ ext/alsa/Makefile ext/arts/Makefile ext/artsd/Makefile ext/audiofile/Makefile +ext/audioresample/Makefile +ext/cairo/Makefile ext/cdaudio/Makefile ext/cdparanoia/Makefile ext/dirac/Makefile diff --git a/ext/Makefile.am b/ext/Makefile.am index 42faa62c..934e0a2c 100644 --- a/ext/Makefile.am +++ b/ext/Makefile.am @@ -34,6 +34,18 @@ else AUDIOFILE_DIR= endif +if USE_AUDIORESAMPLE +AUDIORESAMPLE_DIR=audioresample +else +AUDIORESAMPLE_DIR= +endif + +if USE_CAIRO +CAIRO_DIR=cairo +else +CAIRO_DIR= +endif + if USE_CDAUDIO CDAUDIO_DIR=cdaudio else @@ -383,6 +395,8 @@ SUBDIRS=\ $(ARTS_DIR) \ $(ARTSC_DIR) \ $(AUDIOFILE_DIR) \ + $(AUDIORESAMPLE_DIR) \ + $(CAIRO_DIR) \ $(CDAUDIO_DIR) \ $(CDPARANOIA_DIR) \ $(DIRAC_DIR) \ @@ -446,6 +460,8 @@ DIST_SUBDIRS=\ arts \ artsd \ audiofile \ + audioresample \ + cairo \ cdaudio \ cdparanoia \ dirac \ diff --git a/ext/audioresample/Makefile.am b/ext/audioresample/Makefile.am new file mode 100644 index 00000000..777ff3cf --- /dev/null +++ b/ext/audioresample/Makefile.am @@ -0,0 +1,10 @@ + +plugin_LTLIBRARIES = libgstaudioresample.la + +libgstaudioresample_la_SOURCES = gstaudioresample.c +libgstaudioresample_la_CFLAGS = $(GST_CFLAGS) $(AUDIORESAMPLE_CFLAGS) +libgstaudioresample_la_LIBADD = $(AUDIORESAMPLE_LIBS) +libgstaudioresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) + +noinst_HEADERS = gstaudioresample.h + diff --git a/ext/audioresample/gstaudioresample.c b/ext/audioresample/gstaudioresample.c new file mode 100644 index 00000000..f9b8f656 --- /dev/null +++ b/ext/audioresample/gstaudioresample.c @@ -0,0 +1,435 @@ +/* GStreamer + * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu> + * Copyright (C) 2003,2004 David A. Schleef <ds@schleef.org> + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ +/* Element-Checklist-Version: 5 */ + + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#include <string.h> +#include <math.h> + +/*#define DEBUG_ENABLED */ +#include "gstaudioresample.h" +#include <gst/audio/audio.h> + +GST_DEBUG_CATEGORY_STATIC (audioresample_debug); +#define GST_CAT_DEFAULT audioresample_debug + +/* elementfactory information */ +static GstElementDetails gst_audioresample_details = +GST_ELEMENT_DETAILS ("Audio scaler", + "Filter/Converter/Audio", + "Resample audio", + "David Schleef <ds@schleef.org>"); + +/* Audioresample signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + ARG_0, + ARG_FILTERLEN +}; + +#define SUPPORTED_CAPS \ + GST_STATIC_CAPS (\ + "audio/x-raw-int, " \ + "rate = (int) [ 1, MAX ], " \ + "channels = (int) [ 1, MAX ], " \ + "endianness = (int) BYTE_ORDER, " \ + "width = (int) 16, " \ + "depth = (int) 16, " \ + "signed = (boolean) true") + +#if 0 + /* disabled because it segfaults */ +"audio/x-raw-float, " + "rate = (int) [ 1, MAX ], " + "channels = (int) [ 1, MAX ], " + "endianness = (int) BYTE_ORDER, " "width = (int) 32") +#endif + static GstStaticPadTemplate gst_audioresample_sink_template = + GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS); + + static GstStaticPadTemplate gst_audioresample_src_template = + GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS); + + static void gst_audioresample_base_init (gpointer g_class); + static void gst_audioresample_class_init (AudioresampleClass * klass); + static void gst_audioresample_init (Audioresample * audioresample); + static void gst_audioresample_dispose (GObject * object); + + static void gst_audioresample_chain (GstPad * pad, GstData * _data); + + static void gst_audioresample_set_property (GObject * object, + guint prop_id, const GValue * value, GParamSpec * pspec); + static void gst_audioresample_get_property (GObject * object, + guint prop_id, GValue * value, GParamSpec * pspec); + + static GstElementClass *parent_class = NULL; + +/*static guint gst_audioresample_signals[LAST_SIGNAL] = { 0 }; */ + + GType audioresample_get_type (void) + { + static GType audioresample_type = 0; + + if (!audioresample_type) + { + static const GTypeInfo audioresample_info = { + sizeof (AudioresampleClass), + gst_audioresample_base_init, + NULL, + (GClassInitFunc) gst_audioresample_class_init, + NULL, + NULL, + sizeof (Audioresample), 0, + (GInstanceInitFunc) gst_audioresample_init,}; + + audioresample_type = + g_type_register_static (GST_TYPE_ELEMENT, "Audioresample", + &audioresample_info, 0); + } + return audioresample_type; + } + +static void gst_audioresample_base_init (gpointer g_class) +{ + GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class); + + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_audioresample_src_template)); + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_audioresample_sink_template)); + + gst_element_class_set_details (gstelement_class, &gst_audioresample_details); +} + +static void gst_audioresample_class_init (AudioresampleClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + + gobject_class->set_property = gst_audioresample_set_property; + gobject_class->get_property = gst_audioresample_get_property; + gobject_class->dispose = gst_audioresample_dispose; + + g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FILTERLEN, + g_param_spec_int ("filter_length", "filter_length", "filter_length", + 0, G_MAXINT, 16, G_PARAM_READWRITE | G_PARAM_CONSTRUCT)); + + parent_class = g_type_class_ref (GST_TYPE_ELEMENT); + + GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0, + "audioresample element"); +} + +static void gst_audioresample_expand_caps (GstCaps * caps) +{ + gint i; + + for (i = 0; i < gst_caps_get_size (caps); i++) { + GstStructure *structure = gst_caps_get_structure (caps, i); + const GValue *value; + + value = gst_structure_get_value (structure, "rate"); + if (value == NULL) { + GST_ERROR ("caps structure doesn't have required rate field"); + return; + } + + gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, 0); + } +} + +static GstCaps *gst_audioresample_getcaps (GstPad * pad) +{ + Audioresample *audioresample; + GstCaps *caps; + GstPad *otherpad; + + audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad)); + + otherpad = (pad == audioresample->srcpad) ? audioresample->sinkpad : + audioresample->srcpad; + caps = gst_pad_get_allowed_caps (otherpad); + + gst_audioresample_expand_caps (caps); + + return caps; +} + +static GstCaps *gst_audioresample_fixate (GstPad * pad, const GstCaps * caps) +{ + Audioresample *audioresample; + GstPad *otherpad; + int rate; + GstCaps *copy; + GstStructure *structure; + + audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad)); + + if (pad == audioresample->srcpad) { + otherpad = audioresample->sinkpad; + rate = audioresample->i_rate; + } else + { + otherpad = audioresample->srcpad; + rate = audioresample->o_rate; + } + if (!GST_PAD_IS_NEGOTIATING (otherpad)) + return NULL; + if (gst_caps_get_size (caps) > 1) + return NULL; + + copy = gst_caps_copy (caps); + structure = gst_caps_get_structure (copy, 0); + if (rate) { + if (gst_caps_structure_fixate_field_nearest_int (structure, "rate", rate)) { + return copy; + } + } + gst_caps_free (copy); + return NULL; +} + +static GstPadLinkReturn gst_audioresample_link (GstPad * pad, + const GstCaps * caps) +{ + Audioresample *audioresample; + GstStructure *structure; + int rate; + int channels; + gboolean ret; + GstPad *otherpad; + + audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad)); + + otherpad = (pad == audioresample->srcpad) ? audioresample->sinkpad : + audioresample->srcpad; + + structure = gst_caps_get_structure (caps, 0); + ret = gst_structure_get_int (structure, "rate", &rate); + ret &= gst_structure_get_int (structure, "channels", &channels); + if (!ret) + { + return GST_PAD_LINK_REFUSED; + } + + if (gst_pad_is_negotiated (otherpad)) + { + GstCaps *othercaps = gst_caps_copy (caps); + int otherrate; + GstPadLinkReturn linkret; + + if (pad == audioresample->srcpad) { + otherrate = audioresample->i_rate; + } else { + otherrate = audioresample->o_rate; + } + gst_caps_set_simple (othercaps, "rate", G_TYPE_INT, otherrate, NULL); + linkret = gst_pad_try_set_caps (otherpad, othercaps); + if (GST_PAD_LINK_FAILED (linkret)) { + return GST_PAD_LINK_REFUSED; + } + + } + + audioresample->channels = channels; + resample_set_n_channels (audioresample->resample, audioresample->channels); + if (pad == audioresample->srcpad) { + audioresample->o_rate = rate; + resample_set_output_rate (audioresample->resample, audioresample->o_rate); + GST_DEBUG ("set o_rate to %d", rate); + } else { + audioresample->i_rate = rate; + resample_set_input_rate (audioresample->resample, audioresample->i_rate); + GST_DEBUG ("set i_rate to %d", rate); + } + + return GST_PAD_LINK_OK; +} + +static void gst_audioresample_init (Audioresample * audioresample) +{ + ResampleState *r; + + audioresample->sinkpad = + gst_pad_new_from_template (gst_static_pad_template_get + (&gst_audioresample_sink_template), "sink"); + gst_element_add_pad (GST_ELEMENT (audioresample), audioresample->sinkpad); + gst_pad_set_chain_function (audioresample->sinkpad, gst_audioresample_chain); + gst_pad_set_link_function (audioresample->sinkpad, gst_audioresample_link); + gst_pad_set_getcaps_function (audioresample->sinkpad, + gst_audioresample_getcaps); + gst_pad_set_fixate_function (audioresample->sinkpad, + gst_audioresample_fixate); + + audioresample->srcpad = + gst_pad_new_from_template (gst_static_pad_template_get + (&gst_audioresample_src_template), "src"); + + gst_element_add_pad (GST_ELEMENT (audioresample), audioresample->srcpad); + gst_pad_set_link_function (audioresample->srcpad, gst_audioresample_link); + gst_pad_set_getcaps_function (audioresample->srcpad, + gst_audioresample_getcaps); + gst_pad_set_fixate_function (audioresample->srcpad, gst_audioresample_fixate); + + r = resample_new (); + audioresample->resample = r; + + resample_set_filter_length (r, 64); + resample_set_format (r, RESAMPLE_FORMAT_S16); +} + +static void gst_audioresample_dispose (GObject * object) +{ + Audioresample *audioresample = GST_AUDIORESAMPLE (object); + + if (audioresample->resample) { + resample_free (audioresample->resample); + } + + G_OBJECT_CLASS (parent_class)->dispose (object); +} + +static void gst_audioresample_chain (GstPad * pad, GstData * _data) +{ + GstBuffer *buf = GST_BUFFER (_data); + Audioresample *audioresample; + ResampleState *r; + guchar *data; + gulong size; + int outsize; + GstBuffer *outbuf; + + g_return_if_fail (pad != NULL); + g_return_if_fail (GST_IS_PAD (pad)); + g_return_if_fail (buf != NULL); + + audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad)); + + if (!GST_IS_BUFFER (_data)) { + gst_pad_push (audioresample->srcpad, _data); + return; + } + + if (audioresample->passthru) { + gst_pad_push (audioresample->srcpad, GST_DATA (buf)); + return; + } + + r = audioresample->resample; + + data = GST_BUFFER_DATA (buf); + size = GST_BUFFER_SIZE (buf); + + GST_DEBUG ("got buffer of %ld bytes", size); + + resample_add_input_data (r, data, size, (ResampleCallback) gst_data_unref, + buf); + + outsize = resample_get_output_size (r); + /* FIXME this is audioresample being dumb. dunno why */ + if (outsize == 0) { + GST_ERROR ("overriding outbuf size"); + outsize = size; + } + outbuf = gst_buffer_new_and_alloc (outsize); + + outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize); + GST_BUFFER_SIZE (outbuf) = outsize; + + GST_BUFFER_TIMESTAMP (outbuf) = + audioresample->offset * GST_SECOND / audioresample->o_rate; + audioresample->offset += outsize / sizeof (gint16) / audioresample->channels; + + gst_pad_push (audioresample->srcpad, GST_DATA (outbuf)); +} + +static void + gst_audioresample_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + Audioresample *audioresample; + + /* it's not null if we got it, but it might not be ours */ + g_return_if_fail (GST_IS_AUDIORESAMPLE (object)); + audioresample = GST_AUDIORESAMPLE (object); + + switch (prop_id) { + case ARG_FILTERLEN: + audioresample->filter_length = g_value_get_int (value); + GST_DEBUG_OBJECT (GST_ELEMENT (audioresample), "new filter length %d\n", + audioresample->filter_length); + resample_set_filter_length (audioresample->resample, + audioresample->filter_length); + break; + default:G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void + gst_audioresample_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + Audioresample *audioresample; + + g_return_if_fail (GST_IS_AUDIORESAMPLE (object)); + audioresample = GST_AUDIORESAMPLE (object); + + switch (prop_id) { + case ARG_FILTERLEN: + g_value_set_int (value, audioresample->filter_length); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + + +static gboolean plugin_init (GstPlugin * plugin) +{ + resample_init (); + + if (!gst_element_register (plugin, "audioresample", GST_RANK_PRIMARY, + GST_TYPE_AUDIORESAMPLE)) { + return FALSE; + } + + return TRUE; +} + +GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, + GST_VERSION_MINOR, + "audioresample", + "Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN) diff --git a/ext/audioresample/gstaudioresample.h b/ext/audioresample/gstaudioresample.h new file mode 100644 index 00000000..fc5115da --- /dev/null +++ b/ext/audioresample/gstaudioresample.h @@ -0,0 +1,74 @@ +/* GStreamer + * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu> + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + + +#ifndef __AUDIORESAMPLE_H__ +#define __AUDIORESAMPLE_H__ + + +#include <gst/gst.h> + +#include <audioresample/resample.h> + + +G_BEGIN_DECLS + + +#define GST_TYPE_AUDIORESAMPLE \ + (audioresample_get_type()) +#define GST_AUDIORESAMPLE(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORESAMPLE,Audioresample)) +#define GST_AUDIORESAMPLE_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORESAMPLE,Audioresample)) +#define GST_IS_AUDIORESAMPLE(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORESAMPLE)) +#define GST_IS_AUDIORESAMPLE_CLASS(obj) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORESAMPLE)) + +typedef struct _Audioresample Audioresample; +typedef struct _AudioresampleClass AudioresampleClass; + +struct _Audioresample { + GstElement element; + + GstPad *sinkpad,*srcpad; + + gboolean passthru; + + gint64 offset; + int channels; + + int i_rate; + int o_rate; + int filter_length; + + ResampleState * resample; +}; + +struct _AudioresampleClass { + GstElementClass parent_class; +}; + +GType gst_audioresample_get_type(void); + + +G_END_DECLS + + +#endif /* __AUDIORESAMPLE_H__ */ diff --git a/gst-libs/gst/media-info/media-info-priv.h b/gst-libs/gst/media-info/media-info-priv.h index fa4ed5c4..3c98240c 100644 --- a/gst-libs/gst/media-info/media-info-priv.h +++ b/gst-libs/gst/media-info/media-info-priv.h @@ -23,6 +23,7 @@ #define __GST_MEDIA_INFO_PRIV_H__ #include <gst/gst.h> +#include <glib/gprintf.h> /* debug */ GST_DEBUG_CATEGORY_EXTERN (gst_media_info_debug); |