mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-19 21:08:24 +00:00
Code Issues 1 Projects Releases Wiki Activity

gst/audioscale/gstaudioscale.c: - fix templates to only support S16, it's the only format that works

Browse source 
Original commit message from CVS:
* gst/audioscale/gstaudioscale.c:
- fix templates to only support S16, it's the only format that works
- make caps nego code use try_set_caps_nonfixed and fixation instead
of try_set_caps twice, which is not nice for autopluggers
- change rank to secondary, so autopluggers can pick it up after
audioconvert
This commit is contained in:
Benjamin Otte 2004-08-02 15:55:54 +00:00
parent d1f73b9a75
commit f423b5ff15
2 changed files with 202 additions and 148 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

10
ChangeLog
View file

@ -1,3 +1,12 @@
2004-07-27 Benjamin Otte <otte@gnome.org>
* gst/audioscale/gstaudioscale.c:
- fix templates to only support S16, it's the only format that works
- make caps nego code use try_set_caps_nonfixed and fixation instead
of try_set_caps twice, which is not nice for autopluggers
- change rank to secondary, so autopluggers can pick it up after
audioconvert
2004-08-02 Iain <iain@prettypeople.org> 2004-08-02 Iain <iain@prettypeople.org>
* gst/interleave/interleave.c (interleave_init), * gst/interleave/interleave.c (interleave_init),
@ -279,6 +288,7 @@
* testsuite/alsa/srcstate.c: * testsuite/alsa/srcstate.c:
add test for alsasrc changing state add test for alsasrc changing state
>>>>>>> 1.958
2004-07-27 Zaheer Abbas Merali <zaheerabbas at merali dot org> 2004-07-27 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* gst/silence/gstsilence.c: (gst_silence_init), (gst_silence_link), * gst/silence/gstsilence.c: (gst_silence_init), (gst_silence_link),

340
gst/audioscale/gstaudioscale.c
View file

@ -52,48 +52,53 @@ enum
/* FILL ME */ /* FILL ME */
}; };
static GstStaticPadTemplate gst_audioscale_sink_template = #define SUPPORTED_CAPS \
GST_STATIC_PAD_TEMPLATE ("sink", GST_STATIC_CAPS (\
GST_PAD_SINK, "audio/x-raw-int, " \
GST_PAD_ALWAYS, "rate = (int) [ 1, MAX ], " \
GST_STATIC_CAPS ("audio/x-raw-int, " "channels = (int) [ 1, MAX ], " \
"rate = (int) [ 1, MAX ], " "endianness = (int) BYTE_ORDER, " \
"channels = (int) [ 1, MAX ], " "width = (int) 16, " \
"endianness = (int) BYTE_ORDER, " "depth = (int) 16, " \
"width = (int) { 8, 16, 24, 32 }, " "signed = (boolean) true")
"depth = (int) [ 1, 32 ], " "signed = (boolean) { true, false }") #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_audioscale_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
static GstStaticPadTemplate gst_audioscale_src_template = static GstStaticPadTemplate gst_audioscale_src_template =
GST_STATIC_PAD_TEMPLATE ("src", GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) { 8, 16, 24, 32 }, "
"depth = (int) [ 1, 32 ], " "signed = (boolean) { true, false }")
);
#define GST_TYPE_AUDIOSCALE_METHOD (gst_audioscale_method_get_type()) #define GST_TYPE_AUDIOSCALE_METHOD (gst_audioscale_method_get_type())
static GType static GType gst_audioscale_method_get_type (void)
gst_audioscale_method_get_type (void) {
{ static GType audioscale_method_type = 0;
static GType audioscale_method_type = 0; static GEnumValue audioscale_methods[] =
static GEnumValue audioscale_methods[] = { {
{GST_RESAMPLE_NEAREST, "0", "Nearest"}, {
{GST_RESAMPLE_BILINEAR, "1", "Bilinear"}, GST_RESAMPLE_NEAREST, "0", "Nearest"}
{GST_RESAMPLE_SINC, "2", "Sinc"}, ,
{0, NULL, NULL}, {
}; GST_RESAMPLE_BILINEAR, "1", "Bilinear"}
, {
GST_RESAMPLE_SINC, "2", "Sinc"}
, {
0, NULL, NULL}
,};
if (!audioscale_method_type) { if (!audioscale_method_type) {
audioscale_method_type = g_enum_register_static ("GstAudioscaleMethod", audioscale_method_type = g_enum_register_static ("GstAudioscaleMethod",
audioscale_methods); audioscale_methods);
} }
return audioscale_method_type; return audioscale_method_type;
} }
static void gst_audioscale_base_init (gpointer g_class); static void gst_audioscale_base_init (gpointer g_class);
static void gst_audioscale_class_init (AudioscaleClass * klass); static void gst_audioscale_class_init (AudioscaleClass * klass);
@ -111,23 +116,21 @@ static GstElementClass *parent_class = NULL;
/*static guint gst_audioscale_signals[LAST_SIGNAL] = { 0 }; */ /*static guint gst_audioscale_signals[LAST_SIGNAL] = { 0 }; */
GType GType audioscale_get_type (void)
audioscale_get_type (void)
{ {
static GType audioscale_type = 0; static GType audioscale_type = 0;
if (!audioscale_type) { if (!audioscale_type)
static const GTypeInfo audioscale_info = { {
sizeof (AudioscaleClass), static const GTypeInfo audioscale_info =
gst_audioscale_base_init, {
NULL, sizeof (AudioscaleClass),
(GClassInitFunc) gst_audioscale_class_init, gst_audioscale_base_init,
NULL, NULL,
NULL, (GClassInitFunc) gst_audioscale_class_init,
sizeof (Audioscale), NULL,
0, NULL,
(GInstanceInitFunc) gst_audioscale_init, sizeof (Audioscale), 0, (GInstanceInitFunc) gst_audioscale_init,};
};
audioscale_type = audioscale_type =
g_type_register_static (GST_TYPE_ELEMENT, "Audioscale", g_type_register_static (GST_TYPE_ELEMENT, "Audioscale",
@ -136,8 +139,7 @@ audioscale_get_type (void)
return audioscale_type; return audioscale_type;
} }
static void static void gst_audioscale_base_init (gpointer g_class)
gst_audioscale_base_init (gpointer g_class)
{ {
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class); GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
@ -149,8 +151,7 @@ gst_audioscale_base_init (gpointer g_class)
gst_element_class_set_details (gstelement_class, &gst_audioscale_details); gst_element_class_set_details (gstelement_class, &gst_audioscale_details);
} }
static void static void gst_audioscale_class_init (AudioscaleClass * klass)
gst_audioscale_class_init (AudioscaleClass * klass)
{ {
GObjectClass *gobject_class; GObjectClass *gobject_class;
GstElementClass *gstelement_class; GstElementClass *gstelement_class;
@ -174,22 +175,23 @@ gst_audioscale_class_init (AudioscaleClass * klass)
parent_class = g_type_class_ref (GST_TYPE_ELEMENT); parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
} }
static void static void gst_audioscale_expand_value (GValue * dest, const GValue * src)
gst_audioscale_expand_value (GValue * dest, const GValue * src)
{ {
int rate_min, rate_max; int rate_min, rate_max;
if (G_VALUE_TYPE (src) == G_TYPE_INT || if (G_VALUE_TYPE (src) == G_TYPE_INT ||
G_VALUE_TYPE (src) == GST_TYPE_INT_RANGE) { G_VALUE_TYPE (src) == GST_TYPE_INT_RANGE)
{
if (G_VALUE_TYPE (src) == G_TYPE_INT) { if (G_VALUE_TYPE (src) == G_TYPE_INT) {
rate_min = g_value_get_int (src); rate_min = g_value_get_int (src);
rate_max = rate_min; rate_max = rate_min;
} else { } else
{
rate_min = gst_value_get_int_range_min (src); rate_min = gst_value_get_int_range_min (src);
rate_max = gst_value_get_int_range_max (src); rate_max = gst_value_get_int_range_max (src);
} }
rate_min /= 2; rate_min = (rate_min + 1) / 2;
if (rate_min < 1) if (rate_min < 1)
rate_min = 1; rate_min = 1;
if (rate_max < G_MAXINT / 2) { if (rate_max < G_MAXINT / 2) {
@ -209,14 +211,18 @@ gst_audioscale_expand_value (GValue * dest, const GValue * src)
g_value_init (dest, GST_TYPE_LIST); g_value_init (dest, GST_TYPE_LIST);
for (i = 0; i < gst_value_list_get_size (src); i++) { for (i = 0; i < gst_value_list_get_size (src); i++) {
const GValue *s = gst_value_list_get_value (src, i); const GValue *s = gst_value_list_get_value (src, i);
GValue d = { 0 }; GValue d =
{
0};
int j; int j;
gst_audioscale_expand_value (&d, s); gst_audioscale_expand_value (&d, s);
for (j = 0; j < gst_value_list_get_size (dest); j++) { for (j = 0; j < gst_value_list_get_size (dest); j++) {
const GValue *s2 = gst_value_list_get_value (dest, j); const GValue *s2 = gst_value_list_get_value (dest, j);
GValue d2 = { 0 }; GValue d2 =
{
0};
gst_value_union (&d2, &d, s2); gst_value_union (&d2, &d, s2);
if (G_VALUE_TYPE (&d2) == GST_TYPE_INT_RANGE) { if (G_VALUE_TYPE (&d2) == GST_TYPE_INT_RANGE) {
@ -234,7 +240,9 @@ gst_audioscale_expand_value (GValue * dest, const GValue * src)
if (gst_value_list_get_size (dest) == 1) { if (gst_value_list_get_size (dest) == 1) {
const GValue *s = gst_value_list_get_value (dest, 0); const GValue *s = gst_value_list_get_value (dest, 0);
GValue d = { 0 }; GValue d =
{
0};
gst_value_init_and_copy (&d, s); gst_value_init_and_copy (&d, s);
g_value_unset (dest); g_value_unset (dest);
@ -248,13 +256,36 @@ gst_audioscale_expand_value (GValue * dest, const GValue * src)
GST_ERROR ("unexpected value type"); GST_ERROR ("unexpected value type");
} }
static GstCaps * static void gst_audioscale_expand_caps (GstCaps * caps)
gst_audioscale_getcaps (GstPad * pad) {
gint i;
/* we do this hack, because the audioscale lib doesn't handle
* rate conversions larger than a factor of 2 */
for (i = 0; i < gst_caps_get_size (caps); i++) {
GstStructure *structure = gst_caps_get_structure (caps, i);
const GValue *value;
GValue dest =
{
0};
value = gst_structure_get_value (structure, "rate");
if (value == NULL) {
GST_ERROR ("caps structure doesn't have required rate field");
return;
}
gst_audioscale_expand_value (&dest, value);
gst_structure_set_value (structure, "rate", &dest);
}
}
static GstCaps *gst_audioscale_getcaps (GstPad * pad)
{ {
Audioscale *audioscale; Audioscale *audioscale;
GstCaps *caps; GstCaps *caps;
GstPad *otherpad; GstPad *otherpad;
int i;
audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad)); audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
@ -262,102 +293,118 @@ gst_audioscale_getcaps (GstPad * pad)
audioscale->srcpad; audioscale->srcpad;
caps = gst_pad_get_allowed_caps (otherpad); caps = gst_pad_get_allowed_caps (otherpad);
/* we do this hack, because the audioscale lib doesn't handle gst_audioscale_expand_caps (caps);
* rate conversions larger than a factor of 2 */
for (i = 0; i < gst_caps_get_size (caps); i++) {
GstStructure *structure = gst_caps_get_structure (caps, i);
const GValue *value;
GValue dest = { 0 };
value = gst_structure_get_value (structure, "rate");
if (value == NULL) {
GST_ERROR ("caps structure doesn't have required rate field");
return NULL;
}
gst_audioscale_expand_value (&dest, value);
gst_structure_set_value (structure, "rate", &dest);
}
return caps; return caps;
} }
static GstPadLinkReturn static GstCaps *gst_audioscale_fixate (GstPad * pad, const GstCaps * caps)
gst_audioscale_link (GstPad * pad, const GstCaps * caps) {
Audioscale *audioscale;
gst_resample_t *r;
GstPad *otherpad;
int rate;
GstCaps *copy;
GstStructure *structure;
audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
r = audioscale->gst_resample;
if (pad == audioscale->srcpad)
{
otherpad = audioscale->sinkpad;
rate = r->i_rate;
} else
{
otherpad = audioscale->srcpad;
rate = r->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 (gst_caps_structure_fixate_field_nearest_int (structure, "rate", rate))
return copy;
gst_caps_free (copy);
return NULL;
}
static GstPadLinkReturn gst_audioscale_link (GstPad * pad, const GstCaps * caps)
{ {
Audioscale *audioscale; Audioscale *audioscale;
gst_resample_t *r; gst_resample_t *r;
GstStructure *structure; GstStructure *structure;
int rate; double *rate, *otherrate;
int channels; int temp;
int ret; gboolean ret;
GstPadLinkReturn link_ret; GstPadLinkReturn link_ret;
GstPad *otherpad; GstPad *otherpad;
GstCaps *copy;
audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad)); audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
r = audioscale->gst_resample; r = audioscale->gst_resample;
otherpad = (pad == audioscale->srcpad) ? audioscale->sinkpad if (pad == audioscale->srcpad)
: audioscale->srcpad; {
otherpad = audioscale->sinkpad;
rate = &r->o_rate;
otherrate = &r->i_rate;
} else
{
otherpad = audioscale->srcpad;
rate = &r->i_rate;
otherrate = &r->o_rate;
}
structure = gst_caps_get_structure (caps, 0); structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "rate", &temp);
ret &= gst_structure_get_int (structure, "channels", &r->channels);
g_return_val_if_fail (ret, GST_PAD_LINK_REFUSED);
*rate = temp;
ret = gst_structure_get_int (structure, "rate", &rate); copy = gst_caps_copy (caps);
ret &= gst_structure_get_int (structure, "channels", &channels); gst_audioscale_expand_caps (copy);
link_ret = gst_pad_try_set_caps_nonfixed (otherpad, copy);
link_ret = gst_pad_try_set_caps (otherpad, gst_caps_copy (caps)); if (GST_PAD_LINK_FAILED (link_ret))
if (GST_PAD_LINK_SUCCESSFUL (link_ret)) {
audioscale->passthru = TRUE;
r->channels = channels;
r->i_rate = rate;
r->o_rate = rate;
return link_ret; return link_ret;
}
audioscale->passthru = FALSE;
caps = gst_pad_get_negotiated_caps (otherpad);
if (gst_pad_is_negotiated (otherpad)) { g_return_val_if_fail (caps, GST_PAD_LINK_REFUSED);
GstCaps *trycaps = gst_caps_copy (caps); structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "rate", &temp);
gst_caps_set_simple (trycaps, g_return_val_if_fail (ret, GST_PAD_LINK_REFUSED);
"rate", G_TYPE_INT, *otherrate = temp;
(int) ((pad == audioscale->srcpad) ? r->i_rate : r->o_rate), NULL); if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float")) {
link_ret = gst_pad_try_set_caps (otherpad, trycaps); r->format = GST_RESAMPLE_FLOAT;
if (GST_PAD_LINK_FAILED (link_ret)) {
return link_ret;
}
}
r->channels = channels;
if (pad == audioscale->srcpad) {
r->o_rate = rate;
} else { } else {
r->i_rate = rate; r->format = GST_RESAMPLE_S16;
} }
audioscale->passthru = (r->i_rate == r->o_rate);
gst_resample_reinit (r); gst_resample_reinit (r);
return GST_PAD_LINK_OK; return link_ret;
} }
static void * static void *gst_audioscale_get_buffer (void *priv, unsigned int size)
gst_audioscale_get_buffer (void *priv, unsigned int size)
{ {
Audioscale *audioscale = priv; Audioscale *audioscale = priv;
audioscale->outbuf = gst_buffer_new (); audioscale->outbuf = gst_buffer_new ();
GST_BUFFER_SIZE (audioscale->outbuf) = size; GST_BUFFER_SIZE (audioscale->outbuf) = size;
GST_BUFFER_DATA (audioscale->outbuf) = g_malloc (size); GST_BUFFER_DATA (audioscale->outbuf) = g_malloc (size);
GST_BUFFER_TIMESTAMP (audioscale->outbuf) = GST_BUFFER_TIMESTAMP (audioscale->outbuf) =
audioscale->offset * GST_SECOND / audioscale->gst_resample->o_rate; audioscale->offset * GST_SECOND / audioscale->gst_resample->o_rate;
audioscale->offset += audioscale->offset +=
size / sizeof (gint16) / audioscale->gst_resample->channels; size / sizeof (gint16) / audioscale->gst_resample->channels;
return GST_BUFFER_DATA (audioscale->outbuf); return GST_BUFFER_DATA (audioscale->outbuf);
} }
static void static void gst_audioscale_init (Audioscale * audioscale)
gst_audioscale_init (Audioscale * audioscale)
{ {
gst_resample_t *r; gst_resample_t *r;
@ -368,6 +415,7 @@ gst_audioscale_init (Audioscale * audioscale)
gst_pad_set_chain_function (audioscale->sinkpad, gst_audioscale_chain); gst_pad_set_chain_function (audioscale->sinkpad, gst_audioscale_chain);
gst_pad_set_link_function (audioscale->sinkpad, gst_audioscale_link); gst_pad_set_link_function (audioscale->sinkpad, gst_audioscale_link);
gst_pad_set_getcaps_function (audioscale->sinkpad, gst_audioscale_getcaps); gst_pad_set_getcaps_function (audioscale->sinkpad, gst_audioscale_getcaps);
gst_pad_set_fixate_function (audioscale->sinkpad, gst_audioscale_fixate);
audioscale->srcpad = audioscale->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get gst_pad_new_from_template (gst_static_pad_template_get
@ -376,6 +424,7 @@ gst_audioscale_init (Audioscale * audioscale)
gst_element_add_pad (GST_ELEMENT (audioscale), audioscale->srcpad); gst_element_add_pad (GST_ELEMENT (audioscale), audioscale->srcpad);
gst_pad_set_link_function (audioscale->srcpad, gst_audioscale_link); gst_pad_set_link_function (audioscale->srcpad, gst_audioscale_link);
gst_pad_set_getcaps_function (audioscale->srcpad, gst_audioscale_getcaps); gst_pad_set_getcaps_function (audioscale->srcpad, gst_audioscale_getcaps);
gst_pad_set_fixate_function (audioscale->srcpad, gst_audioscale_fixate);
r = g_new0 (gst_resample_t, 1); r = g_new0 (gst_resample_t, 1);
audioscale->gst_resample = r; audioscale->gst_resample = r;
@ -395,8 +444,7 @@ gst_audioscale_init (Audioscale * audioscale)
/* we will be reinitialized when the G_PARAM_CONSTRUCTs hit */ /* we will be reinitialized when the G_PARAM_CONSTRUCTs hit */
} }
static void static void gst_audioscale_dispose (GObject * object)
gst_audioscale_dispose (GObject * object)
{ {
Audioscale *audioscale = GST_AUDIOSCALE (object); Audioscale *audioscale = GST_AUDIOSCALE (object);
@ -407,8 +455,7 @@ gst_audioscale_dispose (GObject * object)
G_OBJECT_CLASS (parent_class)->dispose (object); G_OBJECT_CLASS (parent_class)->dispose (object);
} }
static void static void gst_audioscale_chain (GstPad * pad, GstData * _data)
gst_audioscale_chain (GstPad * pad, GstData * _data)
{ {
GstBuffer *buf = GST_BUFFER (_data); GstBuffer *buf = GST_BUFFER (_data);
Audioscale *audioscale; Audioscale *audioscale;
@ -439,16 +486,16 @@ gst_audioscale_chain (GstPad * pad, GstData * _data)
} }
static void static void
gst_audioscale_set_property (GObject * object, guint prop_id, gst_audioscale_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec) const GValue * value, GParamSpec * pspec)
{ {
Audioscale *src; Audioscale *src;
gst_resample_t *r; gst_resample_t *r;
/* it's not null if we got it, but it might not be ours */ /* it's not null if we got it, but it might not be ours */
g_return_if_fail (GST_IS_AUDIOSCALE (object)); g_return_if_fail (GST_IS_AUDIOSCALE (object));
src = GST_AUDIOSCALE (object); src = GST_AUDIOSCALE (object);
r = src->gst_resample; r = src->gst_resample;
switch (prop_id) { switch (prop_id) {
case ARG_FILTERLEN: case ARG_FILTERLEN:
@ -456,11 +503,9 @@ gst_audioscale_set_property (GObject * object, guint prop_id,
GST_DEBUG_OBJECT (GST_ELEMENT (src), "new filter length %d\n", GST_DEBUG_OBJECT (GST_ELEMENT (src), "new filter length %d\n",
r->filter_length); r->filter_length);
break; break;
case ARG_METHOD: case ARG_METHOD:r->method = g_value_get_enum (value);
r->method = g_value_get_enum (value);
break; break;
default: default:G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break; break;
} }
@ -468,8 +513,8 @@ gst_audioscale_set_property (GObject * object, guint prop_id,
} }
static void static void
gst_audioscale_get_property (GObject * object, guint prop_id, GValue * value, gst_audioscale_get_property (GObject * object, guint prop_id,
GParamSpec * pspec) GValue * value, GParamSpec * pspec)
{ {
Audioscale *src; Audioscale *src;
gst_resample_t *r; gst_resample_t *r;
@ -491,14 +536,13 @@ gst_audioscale_get_property (GObject * object, guint prop_id, GValue * value,
} }
static gboolean static gboolean plugin_init (GstPlugin * plugin)
plugin_init (GstPlugin * plugin)
{ {
/* load support library */ /* load support library */
if (!gst_library_load ("gstresample")) if (!gst_library_load ("gstresample"))
return FALSE; return FALSE;
if (!gst_element_register (plugin, "audioscale", GST_RANK_NONE, if (!gst_element_register (plugin, "audioscale", GST_RANK_SECONDARY,
GST_TYPE_AUDIOSCALE)) { GST_TYPE_AUDIOSCALE)) {
return FALSE; return FALSE;
} }