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gst-libs/gst/audio/audio.h: remove buffer-frames from audio caps

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Original commit message from CVS:
2004-01-26  Benjamin Otte  <in7y118@public.uni-hamburg.de>

* gst-libs/gst/audio/audio.h:
remove buffer-frames from audio caps
* gst/audioconvert/gstaudioconvert.c:
fix plugin to really work.
This commit is contained in:
Benjamin Otte 2004-01-26 03:54:21 +00:00
parent 7c1c3a173f
commit b552a2d9f0
3 changed files with 153 additions and 267 deletions
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7
ChangeLog
View File

@ -1,3 +1,10 @@
2004-01-26 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst-libs/gst/audio/audio.h:
remove buffer-frames from audio caps
* gst/audioconvert/gstaudioconvert.c:
fix plugin to really work.
2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net> 2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst-libs/gst/mixer/mixer.c: * gst-libs/gst/mixer/mixer.c:

6
gst-libs/gst/audio/audio.h
View File

@ -57,8 +57,7 @@ G_BEGIN_DECLS
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) { 8, 16, 32 }, " \ "width = (int) { 8, 16, 32 }, " \
"depth = (int) [ 1, 32 ], " \ "depth = (int) [ 1, 32 ], " \
"signed = (boolean) { true, false }, " \ "signed = (boolean) { true, false }"
"buffer-frames = (int) [ 1, MAX ]"
/* "standard" int audio is native order, 16 bit stereo. */ /* "standard" int audio is native order, 16 bit stereo. */
@ -69,8 +68,7 @@ G_BEGIN_DECLS
"endianness = (int) BYTE_ORDER, " \ "endianness = (int) BYTE_ORDER, " \
"width = (int) 16, " \ "width = (int) 16, " \
"depth = (int) 16, " \ "depth = (int) 16, " \
"signed = (boolean) true, " \ "signed = (boolean) true"
"buffer-frames = (int) [ 1, MAX]"
#define GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS \ #define GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS \
"audio/x-raw-float, " \ "audio/x-raw-float, " \

407
gst/audioconvert/gstaudioconvert.c
View File

@ -24,28 +24,11 @@
#endif #endif
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/audio/audio.h>
#include <string.h> #include <string.h>
GST_DEBUG_CATEGORY_STATIC (audio_convert_debug); GST_DEBUG_CATEGORY_STATIC (audio_convert_debug);
#define GST_CAT_DEFAULT (audio_convert_debug) #define GST_CAT_DEFAULT (audio_convert_debug)
#if 0
static void
print_caps (GstCaps *caps)
{
GValue v = { 0, };
GValue s = { 0, };
g_value_init (&v, GST_TYPE_CAPS);
g_value_init (&s, G_TYPE_STRING);
g_value_set_boxed (&v, caps);
g_value_transform (&v, &s);
g_print ("%s\n", g_value_get_string (&s));
g_value_unset (&v);
g_value_unset (&s);
}
#endif
/*** DEFINITIONS **************************************************************/ /*** DEFINITIONS **************************************************************/
#define GST_TYPE_AUDIO_CONVERT (gst_audio_convert_get_type()) #define GST_TYPE_AUDIO_CONVERT (gst_audio_convert_get_type())
@ -61,40 +44,28 @@ typedef struct _GstAudioConvertClass GstAudioConvertClass;
/* this struct is a handy way of passing around all the caps info ... */ /* this struct is a handy way of passing around all the caps info ... */
struct _GstAudioConvertCaps { struct _GstAudioConvertCaps {
/* general caps */ /* general caps */
gint endianness; gint endianness;
gint width; gint width;
gint rate; gint rate;
gint channels; gint channels;
gboolean sign;
/* int audio caps */ /* int audio caps */
gint depth; gint depth;
gboolean is_signed;
}; };
struct _GstAudioConvert { struct _GstAudioConvert {
GstElement element; GstElement element;
/* pads */ /* pads */
GstPad * sink; GstPad * sink;
GstPad * src; GstPad * src;
/* properties */ GstAudioConvertCaps srccaps;
gboolean aggressive; GstAudioConvertCaps sinkcaps;
/* caps: 0 = sink, 1 = src, so always convert from 0 to 1 */
gboolean caps_set[2];
GstAudioConvertCaps caps[2];
gint law[2];
gint endian[2];
gint sign[2];
gint depth[2]; /* in BITS */
gint width[2]; /* in BITS */
gint rate[2];
gint channels[2];
/* conversion functions */ /* conversion functions */
GstBuffer * (* convert_internal) (GstAudioConvert *this, GstBuffer *buf); GstBuffer * (* convert_internal) (GstAudioConvert *this, GstBuffer *buf);
}; };
struct _GstAudioConvertClass { struct _GstAudioConvertClass {
@ -113,8 +84,6 @@ static GType gst_audio_convert_get_type (void);
static void gst_audio_convert_base_init (gpointer g_class); static void gst_audio_convert_base_init (gpointer g_class);
static void gst_audio_convert_class_init (GstAudioConvertClass *klass); static void gst_audio_convert_class_init (GstAudioConvertClass *klass);
static void gst_audio_convert_init (GstAudioConvert *audio_convert); static void gst_audio_convert_init (GstAudioConvert *audio_convert);
static void gst_audio_convert_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec);
static void gst_audio_convert_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec);
/* gstreamer functions */ /* gstreamer functions */
static void gst_audio_convert_chain (GstPad *pad, GstData *_data); static void gst_audio_convert_chain (GstPad *pad, GstData *_data);
@ -143,8 +112,10 @@ enum {
ARG_AGGRESSIVE, ARG_AGGRESSIVE,
}; };
static GstElementClass *parent_class = NULL; #define DEBUG_INIT(bla) \
/*static guint gst_audio_convert_signals[LAST_SIGNAL] = { 0 }; */ GST_DEBUG_CATEGORY_INIT (audio_convert_debug, "audioconvert", 0, "audio conversion element");
GST_BOILERPLATE_FULL (GstAudioConvert, gst_audio_convert, GstElement, GST_TYPE_ELEMENT, DEBUG_INIT);
/*** GSTREAMER PROTOTYPES *****************************************************/ /*** GSTREAMER PROTOTYPES *****************************************************/
@ -154,7 +125,13 @@ GST_STATIC_PAD_TEMPLATE (
GST_PAD_SRC, GST_PAD_SRC,
GST_PAD_ALWAYS, GST_PAD_ALWAYS,
GST_STATIC_CAPS ( GST_STATIC_CAPS (
GST_AUDIO_INT_PAD_TEMPLATE_CAPS "audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) { 8, 16, 32 }, " \
"depth = (int) [ 1, 32 ], " \
"signed = (boolean) { true, false }"
) )
); );
@ -164,37 +141,18 @@ GST_STATIC_PAD_TEMPLATE (
GST_PAD_SINK, GST_PAD_SINK,
GST_PAD_ALWAYS, GST_PAD_ALWAYS,
GST_STATIC_CAPS ( GST_STATIC_CAPS (
GST_AUDIO_INT_PAD_TEMPLATE_CAPS "audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) { 8, 16, 32 }, " \
"depth = (int) [ 1, 32 ], " \
"signed = (boolean) { true, false }"
) )
); );
/*** TYPE FUNCTIONS ***********************************************************/ /*** TYPE FUNCTIONS ***********************************************************/
GType
gst_audio_convert_get_type(void) {
static GType audio_convert_type = 0;
if (!audio_convert_type) {
static const GTypeInfo audio_convert_info = {
sizeof(GstAudioConvertClass),
gst_audio_convert_base_init,
NULL,
(GClassInitFunc)gst_audio_convert_class_init,
NULL,
NULL,
sizeof(GstAudioConvert),
0,
(GInstanceInitFunc)gst_audio_convert_init,
};
audio_convert_type = g_type_register_static(GST_TYPE_ELEMENT,
"GstAudioConvert",
&audio_convert_info, 0);
}
GST_DEBUG_CATEGORY_INIT (audio_convert_debug, "audioconvert", 0, "audio conversion element");
return audio_convert_type;
}
static void static void
gst_audio_convert_base_init (gpointer g_class) gst_audio_convert_base_init (gpointer g_class)
{ {
@ -210,59 +168,11 @@ gst_audio_convert_base_init (gpointer g_class)
static void static void
gst_audio_convert_class_init (GstAudioConvertClass *klass) gst_audio_convert_class_init (GstAudioConvertClass *klass)
{ {
GObjectClass *gobject_class; GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GstElementClass *gstelement_class;
gobject_class = (GObjectClass*)klass;
gstelement_class = (GstElementClass*)klass;
parent_class = g_type_class_ref(GST_TYPE_ELEMENT);
gobject_class->set_property = gst_audio_convert_set_property;
gobject_class->get_property = gst_audio_convert_get_property;
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_AGGRESSIVE,
g_param_spec_boolean("aggressive", "aggressive mode",
"if true, tries any possible format before giving up",
FALSE, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
gstelement_class->change_state = gst_audio_convert_change_state; gstelement_class->change_state = gst_audio_convert_change_state;
} }
static GstCaps *
gst_audioconvert_getcaps (GstPad *pad)
{
GstAudioConvert *this;
GstCaps *othercaps;
GstCaps *caps;
GstPad *otherpad;
int i;
GST_DEBUG ("gst_audioconvert_getcaps");
this = GST_AUDIO_CONVERT (gst_pad_get_parent (pad));
otherpad = (pad == this->src) ? this->sink : this->src;
othercaps = gst_pad_get_allowed_caps (otherpad);
GST_DEBUG_CAPS ("othercaps are", othercaps);
caps = gst_caps_copy (othercaps);
for (i = 0; i < gst_caps_get_size (caps); i++) {
GstStructure *structure = gst_caps_get_structure (caps, i);
gst_audio_structure_set_int (structure, GST_AUDIO_FIELD_CHANNELS |
GST_AUDIO_FIELD_ENDIANNESS |
GST_AUDIO_FIELD_WIDTH |
GST_AUDIO_FIELD_DEPTH |
GST_AUDIO_FIELD_SIGNED);
/* FIXME:
* since gst_structure_set doesn't handle lists, we need to do lists
* manually; would be nice if this could be done more easily */
}
return caps;
}
static void static void
gst_audio_convert_init (GstAudioConvert *this) gst_audio_convert_init (GstAudioConvert *this)
{ {
@ -271,7 +181,6 @@ gst_audio_convert_init (GstAudioConvert *this)
gst_static_pad_template_get (&gst_audio_convert_sink_template), "sink"); gst_static_pad_template_get (&gst_audio_convert_sink_template), "sink");
gst_pad_set_getcaps_function (this->sink, gst_audio_convert_getcaps); gst_pad_set_getcaps_function (this->sink, gst_audio_convert_getcaps);
gst_pad_set_link_function (this->sink, gst_audio_convert_link); gst_pad_set_link_function (this->sink, gst_audio_convert_link);
gst_pad_set_getcaps_function (this->sink, gst_audioconvert_getcaps);
gst_element_add_pad (GST_ELEMENT(this), this->sink); gst_element_add_pad (GST_ELEMENT(this), this->sink);
/* srcpad */ /* srcpad */
@ -279,60 +188,20 @@ gst_audio_convert_init (GstAudioConvert *this)
gst_static_pad_template_get (&gst_audio_convert_src_template), "src"); gst_static_pad_template_get (&gst_audio_convert_src_template), "src");
gst_pad_set_getcaps_function (this->src, gst_audio_convert_getcaps); gst_pad_set_getcaps_function (this->src, gst_audio_convert_getcaps);
gst_pad_set_link_function (this->src, gst_audio_convert_link); gst_pad_set_link_function (this->src, gst_audio_convert_link);
gst_pad_set_getcaps_function (this->src, gst_audioconvert_getcaps);
gst_element_add_pad (GST_ELEMENT(this), this->src); gst_element_add_pad (GST_ELEMENT(this), this->src);
gst_pad_set_chain_function(this->sink, gst_audio_convert_chain); gst_pad_set_chain_function(this->sink, gst_audio_convert_chain);
/* clear important variables */ /* clear important variables */
this->caps_set[0] = this->caps_set[1] = FALSE;
this->convert_internal = NULL; this->convert_internal = NULL;
} }
static void
gst_audio_convert_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
{
GstAudioConvert *audio_convert;
/* it's not null if we got it, but it might not be ours */
g_return_if_fail(GST_IS_AUDIO_CONVERT(object));
audio_convert = GST_AUDIO_CONVERT(object);
switch (prop_id) {
case ARG_AGGRESSIVE:
audio_convert->aggressive = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_convert_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
{
GstAudioConvert *audio_convert;
/* it's not null if we got it, but it might not be ours */
g_return_if_fail(GST_IS_AUDIO_CONVERT(object));
audio_convert = GST_AUDIO_CONVERT(object);
switch (prop_id) {
case ARG_AGGRESSIVE:
g_value_set_boolean (value, audio_convert->aggressive);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/*** GSTREAMER FUNCTIONS ******************************************************/ /*** GSTREAMER FUNCTIONS ******************************************************/
static void static void
gst_audio_convert_chain (GstPad *pad, GstData *_data) gst_audio_convert_chain (GstPad *pad, GstData *data)
{ {
GstBuffer *buf = GST_BUFFER (_data); GstBuffer *buf = GST_BUFFER (data);
GstAudioConvert *this; GstAudioConvert *this;
g_return_if_fail (GST_IS_PAD (pad)); g_return_if_fail (GST_IS_PAD (pad));
@ -346,8 +215,12 @@ gst_audio_convert_chain (GstPad *pad, GstData *_data)
return; return;
} }
g_assert(this->caps_set[0] && this->caps_set[1]); g_assert (gst_pad_is_negotiated (this->sink));
if (!gst_pad_is_negotiated (this->src)) {
gst_data_unref (data);
return;
}
/** /**
* Theory of operation: * Theory of operation:
* - convert the format (endianness, signedness, width, depth) to * - convert the format (endianness, signedness, width, depth) to
@ -393,78 +266,87 @@ gst_audio_convert_getcaps (GstPad *pad)
gst_structure_remove_field (structure, "signed"); gst_structure_remove_field (structure, "signed");
} }
caps = gst_caps_intersect (othercaps, gst_pad_get_pad_template_caps (pad)); caps = gst_caps_intersect (othercaps, gst_pad_get_pad_template_caps (pad));
gst_caps_free(othercaps); gst_caps_free (othercaps);
return caps; return caps;
} }
static gboolean
gst_audio_convert_parse_caps (const GstCaps* gst_caps, GstAudioConvertCaps *caps)
{
GstStructure *structure = gst_caps_get_structure (gst_caps, 0);
g_return_val_if_fail (gst_caps_is_fixed (gst_caps), FALSE);
g_return_val_if_fail (caps != NULL, FALSE);
caps->endianness = G_BYTE_ORDER;
if (!gst_structure_get_int (structure, "channels", &caps->channels) ||
!gst_structure_get_boolean (structure, "signed", &caps->sign) ||
!gst_structure_get_int (structure, "depth", &caps->depth) ||
!gst_structure_get_int (structure, "width", &caps->width) ||
!gst_structure_get_int (structure, "rate", &caps->rate) ||
(caps->width != 8 &&
!gst_structure_get_int (structure, "endianness", &caps->endianness))) {
GST_DEBUG ("could not get some values from structure");
return FALSE;
}
return TRUE;
}
static GstPadLinkReturn static GstPadLinkReturn
gst_audio_convert_link (GstPad *pad, const GstCaps *caps) gst_audio_convert_link (GstPad *pad, const GstCaps *caps)
{ {
GstAudioConvert *this; GstAudioConvert *this;
gint nr = 0; GstPad *otherpad;
gint rate, endianness, depth, width, channels; GstAudioConvertCaps ac_caps, other_ac_caps;
gboolean sign; GstCaps *othercaps;
GstStructure *structure; guint i;
gboolean ret; GstPadLinkReturn ret;
g_return_val_if_fail(GST_IS_PAD(pad), GST_PAD_LINK_REFUSED); g_return_val_if_fail(GST_IS_PAD(pad), GST_PAD_LINK_REFUSED);
g_return_val_if_fail(GST_IS_AUDIO_CONVERT(GST_OBJECT_PARENT (pad)), GST_PAD_LINK_REFUSED); g_return_val_if_fail(GST_IS_AUDIO_CONVERT(GST_OBJECT_PARENT (pad)), GST_PAD_LINK_REFUSED);
this = GST_AUDIO_CONVERT(GST_OBJECT_PARENT (pad)); this = GST_AUDIO_CONVERT(GST_OBJECT_PARENT (pad));
otherpad = (pad == this->src ? this->sink : this->src);
/* nr is 0 for sink pad, 1 for src pad */ /* negotiate sinkpad first */
nr = (pad == this->sink) ? 0 : (pad == this->src) ? 1 : -1; if (pad == this->src &&
g_assert (nr > -1); !gst_pad_is_negotiated (this->sink))
return GST_PAD_LINK_DELAYED;
structure = gst_caps_get_structure (caps, 0);
if (!gst_audio_convert_parse_caps (caps, &ac_caps))
ret = gst_structure_get_int (structure, "channels", &channels);
ret &= gst_structure_get_boolean (structure, "signed", &sign);
ret &= gst_structure_get_int (structure, "depth", &depth);
ret &= gst_structure_get_int (structure, "width", &width);
ret &= gst_structure_get_int (structure, "rate", &rate);
endianness = G_BYTE_ORDER;
if (width != 8) {
ret &= gst_structure_get_int (structure, "endianness", &endianness);
}
if (!ret) {
GST_DEBUG ("could not get some values from structure");
return GST_PAD_LINK_REFUSED; return GST_PAD_LINK_REFUSED;
/* try setting our caps on the other side first */
if (gst_pad_try_set_caps (otherpad, caps) >= GST_PAD_LINK_OK) {
this->srccaps = ac_caps;
this->sinkcaps = ac_caps;
return GST_PAD_LINK_OK;
} }
/* we don't convert rate changes, this is done by audioscale */ /* ok, not those - try setting "any" caps */
/* 1 - nr is "the other caps" */ othercaps = gst_pad_get_allowed_caps (otherpad);
if ((this->caps_set[1 - nr]) && for (i = 0; i < gst_caps_get_size (othercaps); i++) {
(rate != this->rate[1 - nr])) { GstStructure *structure = gst_caps_get_structure (othercaps, i);
GstPad *otherpad; gst_structure_set (structure, "rate", G_TYPE_INT, ac_caps.rate, NULL);
GstPadLinkReturn ret; }
ret = gst_pad_try_set_caps (otherpad, othercaps);
otherpad = (nr) ? this->src : this->sink; gst_caps_free (othercaps);
if (gst_pad_is_negotiated (otherpad)) if (ret < GST_PAD_LINK_OK)
{ return ret;
GstCaps *othercaps = gst_caps_copy (othercaps); if (!gst_audio_convert_parse_caps (caps, &other_ac_caps))
return GST_PAD_LINK_REFUSED;
gst_caps_set_simple (othercaps, "rate", G_TYPE_INT, rate, NULL);
/* woohoo, got it */
ret = gst_pad_try_set_caps (otherpad, othercaps); g_return_val_if_fail (gst_audio_convert_parse_caps (
if (GST_PAD_LINK_FAILED (ret)) gst_pad_get_negotiated_caps (otherpad), &other_ac_caps),
{ GST_PAD_LINK_REFUSED);
GST_DEBUG ("could not gst_pad_try_set_caps on otherpad using othercaps"); if (this->sink == pad) {
return GST_PAD_LINK_REFUSED; this->srccaps = other_ac_caps;
} this->sinkcaps = ac_caps;
} } else {
this->rate[1 - nr] = rate; this->srccaps = ac_caps;
this->sinkcaps = other_ac_caps;
} }
GST_DEBUG ("setting caps_set[%d] to TRUE", nr);
this->caps_set[nr] = TRUE;
this->rate[nr] = rate;
this->channels[nr] = channels;
this->sign[nr] = sign;
this->endian[nr] = endianness;
this->depth[nr] = depth;
this->width[nr] = width;
return GST_PAD_LINK_OK; return GST_PAD_LINK_OK;
} }
@ -476,7 +358,6 @@ gst_audio_convert_change_state (GstElement *element)
switch (GST_STATE_TRANSITION (element)) { switch (GST_STATE_TRANSITION (element)) {
case GST_STATE_PAUSED_TO_READY: case GST_STATE_PAUSED_TO_READY:
this->caps_set[0] = this->caps_set[1] = FALSE;
this->convert_internal = NULL; this->convert_internal = NULL;
break; break;
default: default:
@ -497,17 +378,19 @@ static GstBuffer*
gst_audio_convert_get_buffer (GstBuffer *buf, guint size) gst_audio_convert_get_buffer (GstBuffer *buf, guint size)
{ {
GstBuffer *ret; GstBuffer *ret;
GST_LOG ("new buffer of size %u requersted. Current is: data: %p - size: %u - maxsize: %u\n",
size, buf->data, buf->size, buf->maxsize);
if (buf->maxsize >= size && gst_buffer_is_writable (buf)) { if (buf->maxsize >= size && gst_buffer_is_writable (buf)) {
gst_buffer_ref (buf); gst_buffer_ref (buf);
buf->size = size; buf->size = size;
return buf; GST_LOG ("returning same buffer with adjusted values. data: %p - size: %u - maxsize: %u\n",
} else if (buf->maxsize >= size) { buf->data, buf->size, buf->maxsize);
buf = gst_buffer_copy (buf);
buf->size = size;
return buf; return buf;
} else { } else {
g_assert ((ret = gst_buffer_new_and_alloc (size))); g_assert ((ret = gst_buffer_new_and_alloc (size)));
ret->timestamp = buf->timestamp; gst_buffer_stamp (ret, buf);
GST_LOG ("returning new buffer. data: %p - size: %u - maxsize: %u\n",
ret->data, ret->size, ret->maxsize);
return ret; return ret;
} }
} }
@ -537,42 +420,42 @@ gst_audio_convert_buffer_to_default_format (GstAudioConvert *this, GstBuffer *bu
gint32 *dest; gint32 *dest;
guint8 *src; guint8 *src;
if (this->width[0] == 32 && this->depth[0] == 32 && if (this->sinkcaps.width == 32 && this->sinkcaps.depth == 32 &&
this->endian[0] == G_BYTE_ORDER && this->sign[0] == TRUE) this->sinkcaps.endianness == G_BYTE_ORDER && this->sinkcaps.sign == TRUE)
return buf; return buf;
ret = gst_audio_convert_get_buffer (buf, buf->size * 32 / this->width[0]); ret = gst_audio_convert_get_buffer (buf, buf->size * 32 / this->sinkcaps.width);
count = ret->size / 4; count = ret->size / 4;
src = buf->data + (count - 1) * (this->width[0] / 8); src = buf->data + (count - 1) * (this->sinkcaps.width / 8);
dest = (gint32 *) ret->data; dest = (gint32 *) ret->data;
for (i = count - 1; i >= 0; i--) { for (i = count - 1; i >= 0; i--) {
switch (this->width[0]) { switch (this->sinkcaps.width) {
case 8: case 8:
if (this->sign[0]) { if (this->sinkcaps.sign) {
CONVERT_TO (cur, src, gint8, this->sign[0], this->endian[0], GINT8_IDENTITY, GINT8_IDENTITY); CONVERT_TO (cur, src, gint8, this->sinkcaps.sign, this->sinkcaps.endianness, GINT8_IDENTITY, GINT8_IDENTITY);
} else { } else {
CONVERT_TO (cur, src, guint16, this->sign[0], this->endian[0], GUINT8_IDENTITY, GUINT8_IDENTITY); CONVERT_TO (cur, src, guint16, this->sinkcaps.sign, this->sinkcaps.endianness, GUINT8_IDENTITY, GUINT8_IDENTITY);
} }
break; break;
case 16: case 16:
if (this->sign[0]) { if (this->sinkcaps.sign) {
CONVERT_TO (cur, src, gint16, this->sign[0], this->endian[0], GINT16_FROM_LE, GINT16_FROM_BE); CONVERT_TO (cur, src, gint16, this->sinkcaps.sign, this->sinkcaps.endianness, GINT16_FROM_LE, GINT16_FROM_BE);
} else { } else {
CONVERT_TO (cur, src, guint16, this->sign[0], this->endian[0], GUINT16_FROM_LE, GUINT16_FROM_BE); CONVERT_TO (cur, src, guint16, this->sinkcaps.sign, this->sinkcaps.endianness, GUINT16_FROM_LE, GUINT16_FROM_BE);
} }
break; break;
case 32: case 32:
if (this->sign[0]) { if (this->sinkcaps.sign) {
CONVERT_TO (cur, src, gint32, this->sign[0], this->endian[0], GINT32_FROM_LE, GINT32_FROM_BE); CONVERT_TO (cur, src, gint32, this->sinkcaps.sign, this->sinkcaps.endianness, GINT32_FROM_LE, GINT32_FROM_BE);
} else { } else {
CONVERT_TO (cur, src, guint32, this->sign[0], this->endian[0], GUINT32_FROM_LE, GUINT32_FROM_BE); CONVERT_TO (cur, src, guint32, this->sinkcaps.sign, this->sinkcaps.endianness, GUINT32_FROM_LE, GUINT32_FROM_BE);
} }
break; break;
default: default:
g_assert_not_reached (); g_assert_not_reached ();
} }
cur = cur * ((gint64) 1 << (32 - this->depth[0])); cur = cur * ((gint64) 1 << (32 - this->sinkcaps.depth));
cur = CLAMP (cur, -((gint64)1 << 32), (gint64) 0x7FFFFFFF); cur = CLAMP (cur, -((gint64)1 << 32), (gint64) 0x7FFFFFFF);
write = cur; write = cur;
memcpy (&dest[i], &write, 4); memcpy (&dest[i], &write, 4);
@ -582,12 +465,12 @@ gst_audio_convert_buffer_to_default_format (GstAudioConvert *this, GstBuffer *bu
return ret; return ret;
} }
#define POPULATE(format, be_func, le_func) G_STMT_START{ \ #define POPULATE(format, be_func, le_func) G_STMT_START{ \
format val; \ format val; \
format* p = (format *) dest; \ format* p = (format *) dest; \
int_value >>= (32 - this->depth[1]); \ int_value >>= (32 - this->srccaps.depth); \
val = (format) int_value; \ val = (format) int_value; \
switch (this->endian[1]) { \ switch (this->srccaps.endianness) { \
case G_LITTLE_ENDIAN: \ case G_LITTLE_ENDIAN: \
val = le_func (val); \ val = le_func (val); \
break; \ break; \
@ -610,37 +493,36 @@ gst_audio_convert_buffer_from_default_format (GstAudioConvert *this, GstBuffer *
guint count, i; guint count, i;
gint32 *src; gint32 *src;
if (this->width[1] == 32 && this->depth[1] == 32 && if (this->srccaps.width == 32 && this->srccaps.depth == 32 &&
this->endian[1] == G_BYTE_ORDER && this->sign[1] == TRUE) this->srccaps.endianness == G_BYTE_ORDER && this->srccaps.sign == TRUE)
return buf; return buf;
ret = gst_audio_convert_get_buffer (buf, buf->size * this->width[1] / 32); count = buf->size / 4; /* size is undefined after gst_audio_convert_get_buffer! */
ret = gst_audio_convert_get_buffer (buf, buf->size * this->srccaps.width / 32);
dest = ret->data; dest = ret->data;
src = (gint32 *) buf->data; src = (gint32 *) buf->data;
count = ret->size / (this->width[1] / 8);
for (i = 0; i < count; i++) { for (i = 0; i < count; i++) {
gint32 int_value = *src; gint32 int_value = *src;
src++; src++;
switch (this->width[1]) { switch (this->srccaps.width) {
case 8: case 8:
if (this->sign[1]) { if (this->srccaps.sign) {
POPULATE (gint8, GINT8_IDENTITY, GINT8_IDENTITY); POPULATE (gint8, GINT8_IDENTITY, GINT8_IDENTITY);
} else { } else {
POPULATE (guint8, GUINT8_IDENTITY, GUINT8_IDENTITY); POPULATE (guint8, GUINT8_IDENTITY, GUINT8_IDENTITY);
} }
break; break;
case 16: case 16:
if (this->sign[1]) { if (this->srccaps.sign) {
POPULATE (gint16, GINT16_TO_BE, GINT16_TO_LE); POPULATE (gint16, GINT16_TO_BE, GINT16_TO_LE);
} else { } else {
POPULATE (guint16, GUINT16_TO_BE, GUINT16_TO_LE); POPULATE (guint16, GUINT16_TO_BE, GUINT16_TO_LE);
} }
break; break;
case 32: case 32:
if (this->sign[1]) { if (this->srccaps.sign) {
POPULATE (gint32, GINT32_TO_BE, GINT32_TO_LE); POPULATE (gint32, GINT32_TO_BE, GINT32_TO_LE);
} else { } else {
POPULATE (guint32, GUINT32_TO_BE, GUINT32_TO_LE); POPULATE (guint32, GUINT32_TO_BE, GUINT32_TO_LE);
@ -662,15 +544,15 @@ gst_audio_convert_channels (GstAudioConvert *this, GstBuffer *buf)
gint i, count; gint i, count;
guint32 *src, *dest; guint32 *src, *dest;
if (this->channels[0] == this->channels[1]) if (this->sinkcaps.channels == this->srccaps.channels)
return buf; return buf;
count = GST_BUFFER_SIZE (buf) / 4 / this->channels[0]; count = GST_BUFFER_SIZE (buf) / 4 / this->sinkcaps.channels;
ret = gst_audio_convert_get_buffer (buf, count * 4 * this->channels[1]); ret = gst_audio_convert_get_buffer (buf, count * 4 * this->srccaps.channels);
src = (guint32 *) GST_BUFFER_DATA (buf); src = (guint32 *) GST_BUFFER_DATA (buf);
dest = (guint32 *) GST_BUFFER_DATA (ret); dest = (guint32 *) GST_BUFFER_DATA (ret);
if (this->channels[0] > this->channels[1]) { if (this->sinkcaps.channels > this->srccaps.channels) {
for (i = 0; i < count; i++) { for (i = 0; i < count; i++) {
*dest = *src >> 1; *dest = *src >> 1;
src++; src++;
@ -695,7 +577,6 @@ plugin_init (GstPlugin *plugin)
{ {
if (!gst_element_register (plugin, "audioconvert", GST_RANK_NONE, GST_TYPE_AUDIO_CONVERT)) if (!gst_element_register (plugin, "audioconvert", GST_RANK_NONE, GST_TYPE_AUDIO_CONVERT))
return FALSE; return FALSE;
if (!gst_plugin_load ("gstaudio")) return FALSE;
return TRUE; return TRUE;
} }