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dtsdec ported to 0.10
Original commit message from CVS: dtsdec ported to 0.10
This commit is contained in:
parent
6435014c26
commit
03fa6ba8b1
6 changed files with 189 additions and 152 deletions
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@ -1,3 +1,12 @@
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2005-12-14 Edgard Lima <edgard.lima@indt.org.br>
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* configure.ac:
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* ext/Makefile.am:
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* ext/dts/Makefile.am:
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* ext/dts/gstdtsdec.c:
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* ext/dts/gstdtsdec.h:
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dtsdec ported to 0.10
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2005-12-12 Tim-Philipp Müller <tim at centricular dot net>
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* ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_loop):
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@ -392,6 +392,14 @@ else
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AC_SUBST(X_LIBS)
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fi
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dnl *** DTS ***
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translit(dnm, m, l) AM_CONDITIONAL(USE_DTS, true)
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GST_CHECK_FEATURE(DTS, [dts library], dtsdec, [
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GST_CHECK_LIBHEADER(DTS, dts_pic, dts_init, -lm, dts.h, DTS_LIBS="-ldts_pic -lm")
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AC_SUBST(DTS_LIBS)
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])
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dnl *** musepack ***
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translit(dnm, m, l) AM_CONDITIONAL(USE_MUSEPACK, true)
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GST_CHECK_FEATURE(MUSEPACK, [musepackdec], musepack, [
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@ -541,6 +549,7 @@ ext/wavpack/Makefile
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ext/ivorbis/Makefile
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ext/gsm/Makefile
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ext/libmms/Makefile
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ext/dts/Makefile
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ext/musepack/Makefile
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ext/sdl/Makefile
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docs/Makefile
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@ -46,11 +46,11 @@ endif
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DIVX_DIR=
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# endif
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# if USE_DTS
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# DTS_DIR=dts
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# else
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if USE_DTS
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DTS_DIR=dts
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else
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DTS_DIR=
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# endif
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endif
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if USE_FAAC
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FAAC_DIR=faac
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@ -238,6 +238,7 @@ DIST_SUBDIRS= \
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gsm \
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ivorbis \
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libmms \
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dts \
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musepack \
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sdl \
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swfdec \
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@ -1,8 +1,9 @@
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plugin_LTLIBRARIES = libgstdtsdec.la
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libgstdtsdec_la_SOURCES = gstdtsdec.c
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libgstdtsdec_la_CFLAGS = $(GST_CFLAGS)
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libgstdtsdec_la_LIBADD = $(DTS_LIBS)
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libgstdtsdec_la_CFLAGS = $(GST_CFLAGS) $(LIBOIL_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
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libgstdtsdec_la_LIBADD = $(DTS_LIBS) $(LIBOIL_LIBS) $(GST_PLUGINS_BASE_LIBS) \
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-lgstaudio-@GST_MAJORMINOR@
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libgstdtsdec_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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noinst_HEADERS = gstdtsdec.h
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@ -32,6 +32,10 @@
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#include "gstdtsdec.h"
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#include <liboil/liboil.h>
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#include <liboil/liboilcpu.h>
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#include <liboil/liboilfunction.h>
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GST_DEBUG_CATEGORY_STATIC (dtsdec_debug);
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#define GST_CAT_DEFAULT (dtsdec_debug)
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@ -56,19 +60,19 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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#if defined(LIBDTS_FIXED)
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#define DTS_CAPS "audio/x-raw-int, " \
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"endianness = (int) BYTE_ORDER, " \
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"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \
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"signed = (boolean) true, " \
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"width = (int) 16, " \
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"depth = (int) 16"
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#define SAMPLE_WIDTH 16
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#elif defined(LIBDTS_DOUBLE)
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#define DTS_CAPS "audio/x-raw-float, " \
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"endianness = (int) BYTE_ORDER, " \
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"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \
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"width = (int) 64"
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#define SAMPLE_WIDTH 64
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#else
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#define DTS_CAPS "audio/x-raw-float, " \
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"endianness = (int) BYTE_ORDER, " \
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"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \
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"width = (int) 32"
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#define SAMPLE_WIDTH 32
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#endif
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@ -80,11 +84,10 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
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"rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
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);
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static void gst_dtsdec_base_init (GstDtsDecClass * klass);
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static void gst_dtsdec_class_init (GstDtsDecClass * klass);
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static void gst_dtsdec_init (GstDtsDec * dtsdec);
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GST_BOILERPLATE (GstDtsDec, gst_dtsdec, GstElement, GST_TYPE_ELEMENT);
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static void gst_dtsdec_chain (GstPad * pad, GstData * data);
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static gboolean gst_dtsdec_sink_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_dtsdec_chain (GstPad * pad, GstBuffer * buf);
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static GstStateChangeReturn gst_dtsdec_change_state (GstElement * element,
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GstStateChange transition);
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@ -93,39 +96,11 @@ static void gst_dtsdec_set_property (GObject * object, guint prop_id,
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static void gst_dtsdec_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstElementClass *parent_class = NULL;
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/* static guint gst_dtsdec_signals[LAST_SIGNAL] = { 0 }; */
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GType
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gst_dtsdec_get_type (void)
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{
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static GType dtsdec_type = 0;
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if (!dtsdec_type) {
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static const GTypeInfo dtsdec_info = {
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sizeof (GstDtsDecClass),
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(GBaseInitFunc) gst_dtsdec_base_init,
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NULL, (GClassInitFunc) gst_dtsdec_class_init,
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NULL,
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NULL,
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sizeof (GstDtsDec),
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0,
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(GInstanceInitFunc) gst_dtsdec_init,
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};
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dtsdec_type =
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g_type_register_static (GST_TYPE_ELEMENT, "GstDtsDec", &dtsdec_info, 0);
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GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS audio decoder");
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}
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return dtsdec_type;
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}
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static void
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gst_dtsdec_base_init (GstDtsDecClass * klass)
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gst_dtsdec_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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static GstElementDetails gst_dtsdec_details = {
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"DTS audio decoder",
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"Codec/Decoder/Audio",
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_factory));
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gst_element_class_set_details (element_class, &gst_dtsdec_details);
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GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS audio decoder");
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}
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static void
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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guint cpuflags;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_DRC,
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g_param_spec_boolean ("drc", "Dynamic Range Compression",
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"Use Dynamic Range Compression", FALSE, G_PARAM_READWRITE));
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gobject_class->set_property = gst_dtsdec_set_property;
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gobject_class->get_property = gst_dtsdec_get_property;
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gstelement_class->change_state = gst_dtsdec_change_state;
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_DRC,
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g_param_spec_boolean ("drc", "Dynamic Range Compression",
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"Use Dynamic Range Compression", FALSE, G_PARAM_READWRITE));
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oil_init ();
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klass->dts_cpuflags = 0;
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cpuflags = oil_cpu_get_flags ();
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if (cpuflags & OIL_IMPL_FLAG_MMX)
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klass->dts_cpuflags |= MM_ACCEL_X86_MMX;
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if (cpuflags & OIL_IMPL_FLAG_3DNOW)
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klass->dts_cpuflags |= MM_ACCEL_X86_3DNOW;
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if (cpuflags & OIL_IMPL_FLAG_MMXEXT)
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klass->dts_cpuflags |= MM_ACCEL_X86_MMXEXT;
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GST_LOG ("CPU flags: dts=%08x, liboil=%08x", klass->dts_cpuflags, cpuflags);
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}
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static void
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gst_dtsdec_init (GstDtsDec * dtsdec)
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gst_dtsdec_init (GstDtsDec * dtsdec, GstDtsDecClass * g_class)
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{
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GstElement *element = GST_ELEMENT (dtsdec);
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/* create the sink and src pads */
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dtsdec->sinkpad =
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gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT
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(dtsdec), "sink"), "sink");
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gst_pad_new_from_template (gst_static_pad_template_get
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(&sink_factory), "sink");
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gst_pad_set_chain_function (dtsdec->sinkpad, gst_dtsdec_chain);
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gst_element_add_pad (element, dtsdec->sinkpad);
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gst_pad_set_event_function (dtsdec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_dtsdec_sink_event));
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gst_element_add_pad (GST_ELEMENT (dtsdec), dtsdec->sinkpad);
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dtsdec->srcpad =
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gst_pad_new_from_template (gst_element_get_pad_template (element,
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"src"), "src");
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gst_pad_use_explicit_caps (dtsdec->srcpad);
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gst_element_add_pad (element, dtsdec->srcpad);
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gst_pad_new_from_template (gst_static_pad_template_get
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(&src_factory), "src");
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gst_pad_use_fixed_caps (dtsdec->srcpad);
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gst_element_add_pad (GST_ELEMENT (dtsdec), dtsdec->srcpad);
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GST_OBJECT_FLAG_SET (element, GST_ELEMENT_EVENT_AWARE);
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dtsdec->dynamic_range_compression = FALSE;
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}
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@ -268,7 +257,6 @@ gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition ** pos)
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}
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break;
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default:
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/* error */
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g_warning ("dtsdec: invalid flags 0x%x", flags);
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return 0;
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}
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@ -288,9 +276,10 @@ gst_dtsdec_renegotiate (GstDtsDec * dts)
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GstAudioChannelPosition *pos;
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GstCaps *caps = gst_caps_from_string (DTS_CAPS);
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gint channels = gst_dtsdec_channels (dts->using_channels, &pos);
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gboolean result = FALSE;
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if (!channels)
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return FALSE;
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goto done;
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GST_INFO ("dtsdec renegotiate, channels=%d, rate=%d",
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channels, dts->sample_rate);
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@ -301,44 +290,69 @@ gst_dtsdec_renegotiate (GstDtsDec * dts)
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gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
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g_free (pos);
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return gst_pad_set_explicit_caps (dts->srcpad, caps);
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if (!gst_pad_set_caps (dts->srcpad, caps))
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goto done;
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result = TRUE;
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done:
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if (caps) {
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gst_caps_unref (caps);
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}
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return result;
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}
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static void
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gst_dtsdec_handle_event (GstDtsDec * dts, GstEvent * event)
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static gboolean
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gst_dtsdec_sink_event (GstPad * pad, GstEvent * event)
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{
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if (!event) {
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GST_ELEMENT_ERROR (dts, RESOURCE, READ, (NULL), (NULL));
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return;
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}
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GstDtsDec *dtsdec = GST_DTSDEC (gst_pad_get_parent (pad));
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gboolean ret = FALSE;
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GST_LOG ("Handling event of type %d timestamp %llu", GST_EVENT_TYPE (event),
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GST_EVENT_TIMESTAMP (event));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_DISCONTINUOUS:
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{
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case GST_EVENT_NEWSEGMENT:{
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GstFormat format;
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gint64 val;
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if (!gst_event_discont_get_value (event, GST_FORMAT_TIME, &val) ||
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!GST_CLOCK_TIME_IS_VALID (val)) {
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GST_WARNING ("No time discont value in event %p", event);
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gst_event_parse_new_segment (event, NULL, NULL, &format, &val, NULL,
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NULL);
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if (format != GST_FORMAT_TIME || !GST_CLOCK_TIME_IS_VALID (val)) {
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GST_WARNING ("No time in newsegment event %p", event);
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} else {
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dts->current_ts = val;
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dtsdec->current_ts = val;
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}
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if (dtsdec->cache) {
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gst_buffer_unref (dtsdec->cache);
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dtsdec->cache = NULL;
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}
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ret = gst_pad_event_default (pad, event);
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break;
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}
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/* Fallthrough */
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case GST_EVENT_FLUSH:
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if (dts->cache) {
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gst_buffer_unref (dts->cache);
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dts->cache = NULL;
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case GST_EVENT_TAG:
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case GST_EVENT_EOS:{
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ret = gst_pad_event_default (pad, event);
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break;
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}
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case GST_EVENT_FLUSH_START:
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ret = gst_pad_event_default (pad, event);
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break;
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case GST_EVENT_FLUSH_STOP:
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if (dtsdec->cache) {
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gst_buffer_unref (dtsdec->cache);
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dtsdec->cache = NULL;
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}
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ret = gst_pad_event_default (pad, event);
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break;
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default:
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ret = gst_pad_event_default (pad, event);
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break;
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}
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gst_pad_event_default (dts->sinkpad, event);
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gst_object_unref (dtsdec);
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return ret;
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}
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static void
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@ -351,20 +365,19 @@ gst_dtsdec_update_streaminfo (GstDtsDec * dts)
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gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
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GST_TAG_BITRATE, (guint) dts->bit_rate, NULL);
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gst_element_found_tags_for_pad (GST_ELEMENT (dts),
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dts->srcpad, dts->current_ts, taglist);
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gst_element_found_tags_for_pad (GST_ELEMENT (dts), dts->srcpad, taglist);
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}
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static gboolean
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static GstFlowReturn
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gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data,
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guint length, gint flags, gint sample_rate, gint bit_rate)
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{
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gboolean need_renegotiation = FALSE;
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GstClockTime timestamp = 0;
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gint channels, num_blocks;
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GstBuffer *out;
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gint i, s, c, num_c;
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sample_t *samples;
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GstFlowReturn result = GST_FLOW_OK;
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/* go over stream properties, update caps/streaminfo if needed */
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if (dts->sample_rate != sample_rate) {
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@ -385,7 +398,7 @@ gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data,
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if (dts_frame (dts->state, data, &flags, &dts->level, dts->bias)) {
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GST_WARNING ("dts_frame error");
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return FALSE;
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return GST_FLOW_OK;
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}
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channels = flags & (DTS_CHANNEL_MASK | DTS_LFE);
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@ -398,8 +411,10 @@ gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data,
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if (need_renegotiation == TRUE) {
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GST_DEBUG ("dtsdec: sample_rate:%d stream_chans:0x%x using_chans:0x%x",
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dts->sample_rate, dts->stream_channels, dts->using_channels);
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if (!gst_dtsdec_renegotiate (dts))
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return FALSE;
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if (!gst_dtsdec_renegotiate (dts)) {
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GST_ELEMENT_ERROR (dts, CORE, NEGOTIATION, (NULL), (NULL));
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return GST_FLOW_ERROR;
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}
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}
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if (dts->dynamic_range_compression == FALSE) {
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@ -416,14 +431,18 @@ gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data,
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samples = dts_samples (dts->state);
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num_c = gst_dtsdec_channels (dts->using_channels, NULL);
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out = gst_buffer_new_and_alloc ((SAMPLE_WIDTH / 8) * 256 * num_c);
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if (!out) {
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result = gst_pad_alloc_buffer_and_set_caps (dts->srcpad, 0,
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(SAMPLE_WIDTH / 8) * 256 * num_c, GST_PAD_CAPS (dts->srcpad), &out);
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||||
|
||||
if (result != GST_FLOW_OK) {
|
||||
GST_ELEMENT_ERROR (dts, RESOURCE, FAILED, (NULL), ("Out of memory"));
|
||||
return FALSE;
|
||||
goto done;
|
||||
}
|
||||
|
||||
GST_BUFFER_TIMESTAMP (out) = timestamp;
|
||||
GST_BUFFER_TIMESTAMP (out) = dts->current_ts;
|
||||
GST_BUFFER_DURATION (out) = GST_SECOND * 256 / dts->sample_rate;
|
||||
dts->current_ts += GST_BUFFER_DURATION (out);
|
||||
|
||||
/* libdts returns buffers in 256-sample-blocks per channel,
|
||||
* we want interleaved. And we need to copy anyway... */
|
||||
|
@ -436,43 +455,32 @@ gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data,
|
|||
}
|
||||
|
||||
/* push on */
|
||||
gst_pad_push (dts->srcpad, GST_DATA (out));
|
||||
timestamp += GST_SECOND * 256 / dts->sample_rate;
|
||||
result = gst_pad_push (dts->srcpad, out);
|
||||
|
||||
if (result != GST_FLOW_OK) {
|
||||
gst_buffer_unref (out);
|
||||
goto done;
|
||||
}
|
||||
|
||||
|
||||
}
|
||||
|
||||
dts->current_ts = timestamp;
|
||||
return TRUE;
|
||||
done:
|
||||
|
||||
return result;
|
||||
}
|
||||
|
||||
static void
|
||||
gst_dtsdec_chain (GstPad * pad, GstData * _data)
|
||||
static GstFlowReturn
|
||||
gst_dtsdec_chain (GstPad * pad, GstBuffer * buf)
|
||||
{
|
||||
GstDtsDec *dts;
|
||||
guint8 *data;
|
||||
gint64 size;
|
||||
GstBuffer *buf;
|
||||
gint length, flags, sample_rate, bit_rate, frame_length;
|
||||
|
||||
g_return_if_fail (pad != NULL);
|
||||
g_return_if_fail (_data != NULL);
|
||||
GstFlowReturn result = GST_FLOW_OK;
|
||||
|
||||
dts = GST_DTSDEC (gst_pad_get_parent (pad));
|
||||
|
||||
if (GST_IS_EVENT (_data)) {
|
||||
gst_dtsdec_handle_event (dts, GST_EVENT (_data));
|
||||
return;
|
||||
}
|
||||
|
||||
/* merge with cache, if any. Also make sure timestamps match */
|
||||
buf = GST_BUFFER (_data);
|
||||
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
|
||||
dts->current_ts = GST_BUFFER_TIMESTAMP (buf);
|
||||
GST_DEBUG_OBJECT (dts, "Received buffer with ts %" GST_TIME_FORMAT
|
||||
" duration %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
|
||||
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
|
||||
}
|
||||
|
||||
if (dts->cache) {
|
||||
buf = gst_buffer_join (dts->cache, buf);
|
||||
dts->cache = NULL;
|
||||
|
@ -490,8 +498,9 @@ gst_dtsdec_chain (GstPad * pad, GstData * _data)
|
|||
size--;
|
||||
} else if (length <= size) {
|
||||
GST_DEBUG ("Sync: frame size %d", length);
|
||||
if (!gst_dtsdec_handle_frame (dts, data,
|
||||
length, flags, sample_rate, bit_rate)) {
|
||||
result = gst_dtsdec_handle_frame (dts, data, length,
|
||||
flags, sample_rate, bit_rate);
|
||||
if (result != GST_FLOW_OK) {
|
||||
size = 0;
|
||||
break;
|
||||
}
|
||||
|
@ -513,27 +522,23 @@ gst_dtsdec_chain (GstPad * pad, GstData * _data)
|
|||
}
|
||||
|
||||
gst_buffer_unref (buf);
|
||||
gst_object_unref (dts);
|
||||
|
||||
return result;
|
||||
}
|
||||
|
||||
static GstStateChangeReturn
|
||||
gst_dtsdec_change_state (GstElement * element, GstStateChange transition)
|
||||
{
|
||||
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
||||
GstDtsDec *dts = GST_DTSDEC (element);
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_NULL_TO_READY:{
|
||||
GstCPUFlags cpuflags;
|
||||
uint32_t mm_accel = 0;
|
||||
GstDtsDecClass *klass;
|
||||
|
||||
cpuflags = gst_cpu_get_flags ();
|
||||
if (cpuflags & GST_CPU_FLAG_MMX)
|
||||
mm_accel |= MM_ACCEL_X86_MMX;
|
||||
if (cpuflags & GST_CPU_FLAG_3DNOW)
|
||||
mm_accel |= MM_ACCEL_X86_3DNOW;
|
||||
if (cpuflags & GST_CPU_FLAG_MMXEXT)
|
||||
mm_accel |= MM_ACCEL_X86_MMXEXT;
|
||||
|
||||
dts->state = dts_init (mm_accel);
|
||||
klass = GST_DTSDEC_CLASS (G_OBJECT_GET_CLASS (dts));
|
||||
dts->state = dts_init (klass->dts_cpuflags);
|
||||
break;
|
||||
}
|
||||
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
||||
|
@ -548,8 +553,23 @@ gst_dtsdec_change_state (GstElement * element, GstStateChange transition)
|
|||
dts->bias = 0;
|
||||
dts->current_ts = 0;
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
||||
dts->samples = NULL;
|
||||
if (dts->cache) {
|
||||
gst_buffer_unref (dts->cache);
|
||||
dts->cache = NULL;
|
||||
}
|
||||
break;
|
||||
case GST_STATE_CHANGE_READY_TO_NULL:
|
||||
dts_free (dts->state);
|
||||
|
@ -559,10 +579,7 @@ gst_dtsdec_change_state (GstElement * element, GstStateChange transition)
|
|||
break;
|
||||
}
|
||||
|
||||
if (GST_ELEMENT_CLASS (parent_class)->change_state)
|
||||
return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
||||
|
||||
return GST_STATE_CHANGE_SUCCESS;
|
||||
return ret;
|
||||
}
|
||||
|
||||
static void
|
||||
|
@ -600,9 +617,6 @@ gst_dtsdec_get_property (GObject * object, guint prop_id, GValue * value,
|
|||
static gboolean
|
||||
plugin_init (GstPlugin * plugin)
|
||||
{
|
||||
if (!gst_library_load ("gstbytestream") || !gst_library_load ("gstaudio"))
|
||||
return FALSE;
|
||||
|
||||
if (!gst_element_register (plugin, "dtsdec", GST_RANK_PRIMARY,
|
||||
GST_TYPE_DTSDEC))
|
||||
return FALSE;
|
||||
|
|
|
@ -22,7 +22,6 @@
|
|||
#define __GST_DTSDEC_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/bytestream/bytestream.h>
|
||||
|
||||
G_BEGIN_DECLS
|
||||
|
||||
|
@ -41,37 +40,41 @@ typedef struct _GstDtsDec GstDtsDec;
|
|||
typedef struct _GstDtsDecClass GstDtsDecClass;
|
||||
|
||||
struct _GstDtsDec {
|
||||
GstElement element;
|
||||
GstElement element;
|
||||
|
||||
/* pads */
|
||||
GstPad *sinkpad,
|
||||
*srcpad;
|
||||
GstPad *sinkpad,
|
||||
*srcpad;
|
||||
|
||||
/* stream properties */
|
||||
gint bit_rate;
|
||||
gint sample_rate;
|
||||
gint stream_channels;
|
||||
gint request_channels;
|
||||
gint using_channels;
|
||||
gint bit_rate;
|
||||
gint sample_rate;
|
||||
gint stream_channels;
|
||||
gint request_channels;
|
||||
gint using_channels;
|
||||
|
||||
/* decoding properties */
|
||||
sample_t level;
|
||||
sample_t bias;
|
||||
gboolean dynamic_range_compression;
|
||||
sample_t *samples;
|
||||
dts_state_t *state;
|
||||
sample_t level;
|
||||
sample_t bias;
|
||||
gboolean dynamic_range_compression;
|
||||
sample_t *samples;
|
||||
dts_state_t *state;
|
||||
|
||||
/* Data left over from the previous buffer */
|
||||
GstBuffer *cache;
|
||||
GstBuffer *cache;
|
||||
|
||||
/* keep track of time */
|
||||
GstClockTime current_ts;
|
||||
GstClockTime current_ts;
|
||||
};
|
||||
|
||||
struct _GstDtsDecClass {
|
||||
GstElementClass parent_class;
|
||||
|
||||
guint32 dts_cpuflags;
|
||||
};
|
||||
|
||||
GType gst_dtsdec_get_type(void);
|
||||
|
||||
G_END_DECLS
|
||||
|
||||
#endif /* __GST_DTSDEC_H__ */
|
||||
|
|
Loading…
Reference in a new issue