)
])
+dnl *** ivorbis ***
+dnl AM_PATH_IVORBIS only takes two options
+translit(dnm, m, l) AM_CONDITIONAL(USE_IVORBIS, true)
+AG_GST_CHECK_FEATURE(IVORBIS, [integer vorbis plug-in], ivorbisdec, [
+ AG_GST_PKG_CHECK_MODULES(IVORBIS, vorbisidec)
+ if test $HAVE_IVORBIS = no
+ then
+ IVORBIS_LIBS=
+ IVORBIS_CFLAGS=
+ AC_CHECK_LIB(vorbisidec, vorbis_block_init,
+ [IVORBIS_LIBS=-lvorbisidec
+ HAVE_IVORBIS=yes
+ case $host in
+ arm-*-*)
+ IVORBIS_CFLAGS="-D_ARM_ASSEM_ $IVORBIS_CFLAGS"
+ esac
+ ],
+ HAVE_IVORBIS=no)
+ AC_SUBST(IVORBIS_LIBS)
+ AC_SUBST(IVORBIS_CFLAGS)
+ fi
+])
+
dnl *** libgio ***
translit(dnm, m, l) AM_CONDITIONAL(USE_GIO, true)
AG_GST_CHECK_FEATURE(GIO, [GIO library], gio, [
if USE_VORBIS
VORBIS_DIR=vorbis
-else
+endif
+
+if USE_IVORBIS
+VORBIS_DIR=vorbis
+endif
+
+if !USE_VORBIS
+if !USE_IVORBIS
VORBIS_DIR=
endif
+endif
if USE_THEORA
THEORA_DIR=theora
-plugin_LTLIBRARIES = libgstvorbis.la
+plugin_LTLIBRARIES =
+
+if USE_VORBIS
+plugin_LTLIBRARIES += libgstvorbis.la
libgstvorbis_la_SOURCES = gstvorbis.c \
gstvorbisdec.c \
+ gstvorbisdeclib.c \
gstvorbisenc.c \
gstvorbisparse.c \
gstvorbistag.c \
gstvorbiscommon.c
-libgstvorbis_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(VORBIS_CFLAGS)
+libgstvorbis_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(VORBIS_CFLAGS)
## AM_PATH_VORBIS also sets VORBISENC_LIBS
libgstvorbis_la_LIBADD = \
$(top_builddir)/gst-libs/gst/tag/libgsttag-@GST_MAJORMINOR@.la \
$(VORBIS_LIBS) $(VORBISENC_LIBS)
libgstvorbis_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstvorbis_la_LIBTOOLFLAGS = --tag=disable-static
+endif
+
+if USE_IVORBIS
+plugin_LTLIBRARIES += libgstivorbisdec.la
+
+libgstivorbisdec_la_SOURCES = gstivorbisdec.c \
+ gstvorbisdec.c gstvorbisdeclib.c gstvorbiscommon.c
+libgstivorbisdec_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) \
+ -DTREMOR $(IVORBIS_CFLAGS)
+libgstivorbisdec_la_LIBADD = \
+ $(top_builddir)/gst-libs/gst/tag/libgsttag-@GST_MAJORMINOR@.la \
+ $(top_builddir)/gst-libs/gst/audio/libgstaudio-@GST_MAJORMINOR@.la \
+ $(GST_LIBS) $(IVORBIS_LIBS)
+libgstivorbisdec_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstivorbisdec_la_LIBTOOLFLAGS = --tag=disable-static
+endif
noinst_HEADERS = gstvorbisenc.h \
gstvorbisdec.h \
+ gstvorbisdeclib.h \
gstvorbisparse.h \
gstvorbistag.h \
gstvorbiscommon.h
--- /dev/null
+/* 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstvorbisdec.h"
+
+GST_DEBUG_CATEGORY (vorbisdec_debug);
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+
+ /* if tremor is around, there is probably good reason for it, so preferred */
+ if (!gst_element_register (plugin, "ivorbisdec", GST_RANK_PRIMARY + 1,
+ gst_vorbis_dec_get_type ()))
+ return FALSE;
+
+ GST_DEBUG_CATEGORY_INIT (vorbisdec_debug, "ivorbisdec", 0,
+ "vorbis decoding element (integer decoder)");
+
+ return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+ GST_VERSION_MINOR,
+ "ivorbisdec",
+ "Vorbis Tremor decoder",
+ plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
#define GST_CAT_DEFAULT vorbisdec_debug
static const GstElementDetails vorbis_dec_details =
-GST_ELEMENT_DETAILS ("Vorbis audio decoder",
- "Codec/Decoder/Audio",
- "decode raw vorbis streams to float audio",
- "Benjamin Otte <in7y118@public.uni-hamburg.de>");
+ GST_VORBIS_DEC_ELEMENT_DETAILS;
static GstStaticPadTemplate vorbis_dec_src_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-float, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, 256 ], " "endianness = (int) BYTE_ORDER, "
-/* no ifdef in macros, please
-#ifdef GST_VORBIS_DEC_SEQUENTIAL
- "layout = \"sequential\", "
-#endif
-*/
- "width = (int) 32")
- );
+ GST_VORBIS_DEC_SRC_CAPS);
static GstStaticPadTemplate vorbis_dec_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_STATIC_CAPS ("audio/x-vorbis")
);
-GST_BOILERPLATE (GstVorbisDec, gst_vorbis_dec, GstElement, GST_TYPE_ELEMENT);
+GST_BOILERPLATE (GST_VORBIS_DEC_GLIB_TYPE_NAME, gst_vorbis_dec, GstElement,
+ GST_TYPE_ELEMENT);
static void vorbis_dec_finalize (GObject * object);
static gboolean vorbis_dec_sink_event (GstPad * pad, GstEvent * event);
{
GstCaps *caps;
const GstAudioChannelPosition *pos = NULL;
+ gint width = GST_VORBIS_DEC_DEFAULT_SAMPLE_WIDTH;
switch (vd->vi.channels) {
case 1:
}
}
- vd->width = 4;
+ /* negotiate width with downstream */
+ caps = gst_pad_get_allowed_caps (vd->srcpad);
+ if (caps) {
+ if (!gst_caps_is_empty (caps)) {
+ GstStructure *s;
+
+ s = gst_caps_get_structure (caps, 0);
+ /* template ensures 16 or 32 */
+ gst_structure_get_int (s, "width", &width);
+ }
+ gst_caps_unref (caps);
+ }
+ vd->width = width >> 3;
caps = gst_caps_copy (gst_pad_get_pad_template_caps (vd->srcpad));
gst_caps_set_simple (caps, "rate", G_TYPE_INT, vd->vi.rate,
- "channels", G_TYPE_INT, vd->vi.channels);
+ "channels", G_TYPE_INT, vd->vi.channels,
+ "width", G_TYPE_INT, width, NULL);
if (pos) {
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
GST_DEBUG_OBJECT (vd, "parsing comment packet");
buf = gst_buffer_new ();
- GST_BUFFER_DATA (buf) = packet->packet;
- GST_BUFFER_SIZE (buf) = packet->bytes;
+ GST_BUFFER_DATA (buf) = gst_ogg_packet_data (packet);
+ GST_BUFFER_SIZE (buf) = gst_ogg_packet_size (packet);
list =
gst_tag_list_from_vorbiscomment_buffer (buf, (guint8 *) "\003vorbis", 7,
GST_DEBUG_OBJECT (vd, "parsing header packet");
/* Packetno = 0 if the first byte is exactly 0x01 */
- packet->b_o_s = (packet->packet[0] == 0x1) ? 1 : 0;
+ packet->b_o_s = ((gst_ogg_packet_data (packet))[0] == 0x1) ? 1 : 0;
if ((ret = vorbis_synthesis_headerin (&vd->vi, &vd->vc, packet)))
goto header_read_error;
- switch (packet->packet[0]) {
+ switch ((gst_ogg_packet_data (packet))[0]) {
case 0x01:
res = vorbis_handle_identification_packet (vd);
break;
}
}
-/* These samples can be outside of the float -1.0 -- 1.0 range, this
- * is allowed, downstream elements are supposed to clip */
-static void
-copy_samples (float *out, float **in, guint samples, gint channels)
-{
- gint i, j;
-
-#ifdef GST_VORBIS_DEC_SEQUENTIAL
- for (i = 0; i < channels; i++) {
- memcpy (out, in[i], samples * sizeof (float));
- out += samples;
- }
-#else
- for (j = 0; j < samples; j++) {
- for (i = 0; i < channels; i++) {
- *out++ = in[i][j];
- }
- }
-#endif
-}
-
static GstFlowReturn
vorbis_dec_push_forward (GstVorbisDec * dec, GstBuffer * buf)
{
vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet,
GstClockTime timestamp, GstClockTime duration)
{
- float **pcm;
+ vorbis_sample_t **pcm;
guint sample_count;
GstBuffer *out;
GstFlowReturn result;
goto wrong_samples;
/* copy samples in buffer */
- copy_samples ((float *) GST_BUFFER_DATA (out), pcm, sample_count,
- vd->vi.channels);
+ copy_samples ((vorbis_sample_t *) GST_BUFFER_DATA (out), pcm, sample_count,
+ vd->vi.channels, vd->width);
GST_LOG_OBJECT (vd, "setting output size to %d", size);
GST_BUFFER_SIZE (out) = size;
static GstFlowReturn
vorbis_dec_decode_buffer (GstVorbisDec * vd, GstBuffer * buffer)
{
- ogg_packet packet;
+ ogg_packet *packet;
+ ogg_packet_wrapper packet_wrapper;
GstFlowReturn result = GST_FLOW_OK;
/* make ogg_packet out of the buffer */
- packet.packet = GST_BUFFER_DATA (buffer);
- packet.bytes = GST_BUFFER_SIZE (buffer);
- packet.granulepos = -1;
- packet.packetno = 0; /* we don't care */
+ gst_ogg_packet_wrapper_from_buffer (&packet_wrapper, buffer);
+ packet = gst_ogg_packet_from_wrapper (&packet_wrapper);
+ /* set some more stuff */
+ packet->granulepos = -1;
+ packet->packetno = 0; /* we don't care */
/* EOS does not matter, it is used in vorbis to implement clipping the last
* block of samples based on the granulepos. We clip based on segments. */
- packet.e_o_s = 0;
+ packet->e_o_s = 0;
- GST_LOG_OBJECT (vd, "decode buffer of size %ld", packet.bytes);
+ GST_LOG_OBJECT (vd, "decode buffer of size %ld", packet->bytes);
/* error out on empty header packets, but just skip empty data packets */
- if (G_UNLIKELY (packet.bytes == 0)) {
+ if (G_UNLIKELY (packet->bytes == 0)) {
if (vd->initialized)
goto empty_buffer;
else
}
/* switch depending on packet type */
- if (packet.packet[0] & 1) {
+ if ((gst_ogg_packet_data (packet))[0] & 1) {
if (vd->initialized) {
GST_WARNING_OBJECT (vd, "Already initialized, so ignoring header packet");
goto done;
}
- result = vorbis_handle_header_packet (vd, &packet);
+ result = vorbis_handle_header_packet (vd, packet);
} else {
GstClockTime timestamp, duration;
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
- result = vorbis_handle_data_packet (vd, &packet, timestamp, duration);
+ result = vorbis_handle_data_packet (vd, packet, timestamp, duration);
}
done:
#include <gst/gst.h>
-#include <vorbis/codec.h>
+#include "gstvorbisdeclib.h"
G_BEGIN_DECLS
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2010 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+ * Copyright (C) 2010 Nokia Corporation. All rights reserved.
+ * Contact: Stefan Kost <stefan.kost@nokia.com>
+ *
+ * Tremor modifications <2006>:
+ * Chris Lord, OpenedHand Ltd. <chris@openedhand.com>, http://www.o-hand.com/
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstvorbisdeclib.h"
+
+#ifndef TREMOR
+/* These samples can be outside of the float -1.0 -- 1.0 range, this
+ * is allowed, downstream elements are supposed to clip */
+void
+copy_samples (vorbis_sample_t * out, vorbis_sample_t ** in, guint samples,
+ gint channels, gint width)
+{
+ gint i, j;
+
+ g_assert (width == 4);
+
+#ifdef GST_VORBIS_DEC_SEQUENTIAL
+ for (i = 0; i < channels; i++) {
+ memcpy (out, in[i], samples * sizeof (float));
+ out += samples;
+ }
+#else
+ for (j = 0; j < samples; j++) {
+ for (i = 0; i < channels; i++) {
+ *out++ = in[i][j];
+ }
+ }
+#endif
+}
+
+#else
+
+/* Taken from Tremor, misc.h */
+#ifdef _ARM_ASSEM_
+static inline ogg_int32_t
+CLIP_TO_15 (ogg_int32_t x)
+{
+ int tmp;
+ asm volatile ("subs %1, %0, #32768\n\t"
+ "movpl %0, #0x7f00\n\t"
+ "orrpl %0, %0, #0xff\n"
+ "adds %1, %0, #32768\n\t"
+ "movmi %0, #0x8000":"+r" (x), "=r" (tmp)
+ ::"cc");
+
+ return (x);
+}
+#else
+static inline ogg_int32_t
+CLIP_TO_15 (ogg_int32_t x)
+{
+ int ret = x;
+
+ ret -= ((x <= 32767) - 1) & (x - 32767);
+ ret -= ((x >= -32768) - 1) & (x + 32768);
+ return (ret);
+}
+#endif
+
+static void
+copy_samples_32 (gint32 * out, ogg_int32_t ** in, guint samples, gint channels)
+{
+ gint i, j;
+
+ for (j = 0; j < samples; j++) {
+ for (i = 0; i < channels; i++) {
+ *out++ = CLIP_TO_15 (in[i][j] >> 9);
+ }
+ }
+}
+
+static void
+copy_samples_16 (gint16 * out, ogg_int32_t ** in, guint samples, gint channels)
+{
+ gint i, j;
+
+ for (j = 0; j < samples; j++) {
+ for (i = 0; i < channels; i++) {
+ *out++ = CLIP_TO_15 (in[i][j] >> 9);
+ }
+ }
+}
+
+void
+copy_samples (vorbis_sample_t * out, vorbis_sample_t ** in, guint samples,
+ gint channels, gint width)
+{
+ if (width == 4) {
+ copy_samples_32 ((gint32 *) out, in, samples, channels);
+ } else if (width == 2) {
+ copy_samples_16 ((gint16 *) out, in, samples, channels);
+ } else {
+ g_assert_not_reached ();
+ }
+}
+
+#endif
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2010 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+ * Copyright (C) 2010 Nokia Corporation. All rights reserved.
+ * Contact: Stefan Kost <stefan.kost@nokia.com>
+ *
+ * Tremor modifications <2006>:
+ * Chris Lord, OpenedHand Ltd. <chris@openedhand.com>, http://www.o-hand.com/
+ *
+ * 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 __GST_VORBIS_DEC_LIB_H__
+#define __GST_VORBIS_DEC_LIB_H__
+
+#include <gst/gst.h>
+
+#ifndef TREMOR
+
+#include <vorbis/codec.h>
+
+typedef float vorbis_sample_t;
+typedef ogg_packet ogg_packet_wrapper;
+
+#define GST_VORBIS_DEC_ELEMENT_DETAILS \
+GST_ELEMENT_DETAILS ("Vorbis audio decoder", \
+ "Codec/Decoder/Audio", \
+ "decode raw vorbis streams to float audio", \
+ "Benjamin Otte <in7y118@public.uni-hamburg.de>")
+
+#define GST_VORBIS_DEC_SRC_CAPS \
+ GST_STATIC_CAPS ("audio/x-raw-float, " "rate = (int) [ 1, MAX ], " \
+ "channels = (int) [ 1, 256 ], " "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 32")
+
+#define GST_VORBIS_DEC_DEFAULT_SAMPLE_WIDTH (32)
+
+#define GST_VORBIS_DEC_GLIB_TYPE_NAME GstVorbisDec
+
+static inline guint8 *
+gst_ogg_packet_data (ogg_packet * p)
+{
+ return (guint8 *) p->packet;
+}
+
+static inline gint
+gst_ogg_packet_size (ogg_packet * p)
+{
+ return p->bytes;
+}
+
+static inline void
+gst_ogg_packet_wrapper_from_buffer (ogg_packet * packet, GstBuffer * buffer)
+{
+ packet->packet = GST_BUFFER_DATA (buffer);
+ packet->bytes = GST_BUFFER_SIZE (buffer);
+}
+
+static inline ogg_packet *
+gst_ogg_packet_from_wrapper (ogg_packet_wrapper * packet)
+{
+ return packet;
+}
+
+#else
+
+#include <tremor/ivorbiscodec.h>
+
+typedef ogg_int32_t vorbis_sample_t;
+typedef struct _ogg_packet_wrapper ogg_packet_wrapper;
+
+struct _ogg_packet_wrapper {
+ ogg_packet packet;
+ ogg_reference ref;
+ ogg_buffer buf;
+};
+
+#define GST_VORBIS_DEC_ELEMENT_DETAILS \
+GST_ELEMENT_DETAILS ("Vorbis audio decoder", \
+ "Codec/Decoder/Audio", \
+ "decode raw vorbis streams to integer audio", \
+ "Benjamin Otte <in7y118@public.uni-hamburg.de>\n" \
+ "Chris Lord <chris@openedhand.com>")
+
+#define GST_VORBIS_DEC_SRC_CAPS \
+ GST_STATIC_CAPS ("audio/x-raw-int, " \
+ "rate = (int) [ 1, MAX ], " \
+ "channels = (int) [ 1, 6 ], " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) { 16, 32 }, " \
+ "depth = (int) 16, " "signed = (boolean) true")
+
+#define GST_VORBIS_DEC_DEFAULT_SAMPLE_WIDTH (16)
+
+/* we need a different type name here */
+#define GST_VORBIS_DEC_GLIB_TYPE_NAME GstIVorbisDec
+
+/* and still have it compile */
+typedef struct _GstVorbisDec GstIVorbisDec;
+typedef struct _GstVorbisDecClass GstIVorbisDecClass;
+
+/* compensate minor variation */
+#define vorbis_synthesis(a, b) vorbis_synthesis (a, b, 1)
+
+static inline guint8 *
+gst_ogg_packet_data (ogg_packet * p)
+{
+ return (guint8 *) p->packet->buffer->data;
+}
+
+static inline gint
+gst_ogg_packet_size (ogg_packet * p)
+{
+ return p->packet->buffer->size;
+}
+
+static inline void
+gst_ogg_packet_wrapper_from_buffer (ogg_packet_wrapper * packet,
+ GstBuffer * buffer)
+{
+ ogg_reference *ref = &packet->ref;
+ ogg_buffer *buf = &packet->buf;
+
+ buf->data = GST_BUFFER_DATA (buffer);
+ buf->size = GST_BUFFER_SIZE (buffer);
+ buf->refcount = 1;
+ buf->ptr.owner = NULL;
+ buf->ptr.next = NULL;
+
+ ref->buffer = buf;
+ ref->begin = 0;
+ ref->length = buf->size;
+ ref->next = NULL;
+
+ packet->packet.packet = ref;
+ packet->packet.bytes = ref->length;
+}
+
+static inline ogg_packet *
+gst_ogg_packet_from_wrapper (ogg_packet_wrapper * packet)
+{
+ return &(packet->packet);
+}
+
+#endif
+
+void copy_samples (vorbis_sample_t *out, vorbis_sample_t **in,
+ guint samples, gint channels, gint width);
+
+
+#endif /* __GST_VORBIS_DEC_LIB_H__ */
projects / platform / upstream / gstreamer.git / commitdiff