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fix up audioconvert caps nego, remove float stuff, remove rate stuff gst-launch-0.7 -v sinesrc ! audioconvert ! audi...

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Original commit message from CVS:

fix up audioconvert caps nego, remove float stuff, remove rate stuff
gst-launch-0.7  -v sinesrc ! audioconvert ! audio/x-raw-int,rate=23000 ! wavenc ! filesink location=test.wav now writes a completely useless 23000 Hz wave file
This commit is contained in:
Thomas Vander Stichele 2004-01-12 19:46:45 +00:00
parent 5c63b06853
commit 2a415f1abd
2 changed files with 93 additions and 159 deletions
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15
ChangeLog
View file

@ -1,3 +1,18 @@
2004-01-12 Thomas Vander Stichele <thomas at apestaart dot org>
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_get_type),
(gst_audio_convert_class_init), (gst_audioconvert_getcaps),
(gst_audio_convert_init), (gst_audio_convert_set_property),
(gst_audio_convert_get_property), (gst_audio_convert_chain),
(gst_audio_convert_link),
(gst_audio_convert_buffer_to_default_format),
(gst_audio_convert_buffer_from_default_format), (plugin_init):
- implement _getcaps and use it
- improve linking
- remove float caps since no float conversion is actually done
- remove properties and arguments that were to be used for rate
conversion
2004-01-12 Thomas Vander Stichele <thomas at apestaart dot org>
* gst-libs/gst/audio/audio.c: (_gst_audio_structure_set_list),

237
gst/audioconvert/gstaudioconvert.c
View file

@ -27,6 +27,9 @@
#include <gst/audio/audio.h>
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (audio_convert_debug);
#define GST_CAT_DEFAULT (audio_convert_debug)
#if 0
static void
print_caps (GstCaps *caps)
@ -63,14 +66,9 @@ struct _GstAudioConvertCaps {
gint rate;
gint channels;
gboolean is_float; /* true iff a pad is carrying float data */
/* int audio caps */
gint depth;
gboolean is_signed;
/* float audio caps */
guint buffer_frames;
};
struct _GstAudioConvert {
@ -82,7 +80,6 @@ struct _GstAudioConvert {
/* properties */
gboolean aggressive;
guint min_rate, max_rate, rate_steps;
/* caps: 0 = sink, 1 = src, so always convert from 0 to 1 */
gboolean caps_set[2];
@ -92,7 +89,7 @@ struct _GstAudioConvert {
gint endian[2];
gint sign[2];
gint depth[2]; /* in BITS */
gint width[2]; /* in BYTES */
gint width[2]; /* in BITS */
gint rate[2];
gint channels[2];
@ -143,9 +140,6 @@ enum {
enum {
ARG_0,
ARG_AGGRESSIVE,
ARG_MIN_RATE,
ARG_MAX_RATE,
ARG_RATE_STEPS
};
static GstElementClass *parent_class = NULL;
@ -159,8 +153,7 @@ GST_STATIC_PAD_TEMPLATE (
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (
GST_AUDIO_INT_PAD_TEMPLATE_CAPS "; "
GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS
GST_AUDIO_INT_PAD_TEMPLATE_CAPS
)
);
@ -170,8 +163,7 @@ GST_STATIC_PAD_TEMPLATE (
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (
GST_AUDIO_INT_PAD_TEMPLATE_CAPS "; "
GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS
GST_AUDIO_INT_PAD_TEMPLATE_CAPS
)
);
@ -197,6 +189,8 @@ gst_audio_convert_get_type(void) {
"GstAudioConvert",
&audio_convert_info, 0);
}
GST_DEBUG_CATEGORY_INIT (audio_convert_debug, "audioconvert", 0, "audio conversion element");
return audio_convert_type;
}
@ -231,24 +225,43 @@ gst_audio_convert_class_init (GstAudioConvertClass *klass)
"if true, tries any possible format before giving up",
FALSE, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_MIN_RATE,
g_param_spec_uint("min-rate", "minimum rate allowed",
"defines the lower bound for the audio rate",
0, G_MAXUINT, 8000, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_MAX_RATE,
g_param_spec_uint("max-rate", "maximum rate allowed",
"defines the upper bound for the audio rate",
0, G_MAXUINT, 192000, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_RATE_STEPS,
g_param_spec_uint("rate-steps", "rate search steps",
"the number of steps used for searching between min and max rates",
0, G_MAXUINT, 32, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
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
gst_audio_convert_init (GstAudioConvert *this)
{
@ -256,12 +269,14 @@ gst_audio_convert_init (GstAudioConvert *this)
this->sink = gst_pad_new_from_template (
gst_static_pad_template_get (&gst_audio_convert_sink_template), "sink");
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);
/* srcpad */
this->src = gst_pad_new_from_template (
gst_static_pad_template_get (&gst_audio_convert_src_template), "src");
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_pad_set_chain_function(this->sink, gst_audio_convert_chain);
@ -283,15 +298,6 @@ gst_audio_convert_set_property (GObject *object, guint prop_id, const GValue *va
case ARG_AGGRESSIVE:
audio_convert->aggressive = g_value_get_boolean (value);
break;
case ARG_MIN_RATE:
audio_convert->min_rate = g_value_get_uint (value);
break;
case ARG_MAX_RATE:
audio_convert->max_rate = g_value_get_uint (value);
break;
case ARG_RATE_STEPS:
audio_convert->rate_steps = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -311,15 +317,6 @@ gst_audio_convert_get_property (GObject *object, guint prop_id, GValue *value, G
case ARG_AGGRESSIVE:
g_value_set_boolean (value, audio_convert->aggressive);
break;
case ARG_MIN_RATE:
g_value_set_uint (value, audio_convert->min_rate);
break;
case ARG_MAX_RATE:
g_value_set_uint (value, audio_convert->max_rate);
break;
case ARG_RATE_STEPS:
g_value_set_uint (value, audio_convert->rate_steps);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -335,10 +332,10 @@ gst_audio_convert_chain (GstPad *pad, GstData *_data)
GstBuffer *buf = GST_BUFFER (_data);
GstAudioConvert *this;
g_return_if_fail(GST_IS_PAD(pad));
g_return_if_fail(buf != NULL);
g_return_if_fail(GST_IS_AUDIO_CONVERT(GST_OBJECT_PARENT (pad)));
this = GST_AUDIO_CONVERT(GST_OBJECT_PARENT (pad));
g_return_if_fail (GST_IS_PAD (pad));
g_return_if_fail (buf != NULL);
g_return_if_fail (GST_IS_AUDIO_CONVERT (GST_OBJECT_PARENT (pad)));
this = GST_AUDIO_CONVERT (GST_OBJECT_PARENT (pad));
/* FIXME */
if (GST_IS_EVENT (buf)) {
@ -379,6 +376,7 @@ gst_audio_convert_link (GstPad *pad, const GstCaps *caps)
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));
/* nr is 0 for sink pad, 1 for src pad */
nr = (pad == this->sink) ? 0 : (pad == this->src) ? 1 : -1;
g_assert (nr > -1);
@ -395,36 +393,42 @@ gst_audio_convert_link (GstPad *pad, const GstCaps *caps)
}
if (!ret) {
GST_DEBUG ("could not get some values from structure");
return GST_PAD_LINK_REFUSED;
}
/* we can't convert rate changes yet */
/* we don't convert rate changes, this is done by audioscale */
/* 1 - nr is "the other caps" */
if ((this->caps_set[1 - nr]) &&
(rate != this->rate[1 - nr])) {
GstPad *otherpad;
GstCaps *othercaps;
GstPadLinkReturn ret;
otherpad = (nr) ? this->src : this->sink;
if (gst_pad_is_negotiated (otherpad)) {
othercaps = gst_caps_copy (gst_pad_get_negotiated_caps (otherpad));
if (gst_pad_is_negotiated (otherpad))
{
GstCaps *othercaps = gst_caps_copy (othercaps);
gst_caps_set_simple (othercaps, "rate", G_TYPE_INT, rate, NULL);
ret = gst_pad_try_set_caps (otherpad, othercaps);
if (GST_PAD_LINK_FAILED (ret)) return ret;
this->rate[1 - nr] = rate;
if (GST_PAD_LINK_FAILED (ret))
{
GST_DEBUG ("could not gst_pad_try_set_caps on otherpad using othercaps");
return GST_PAD_LINK_REFUSED;
}
}
this->rate[1 - nr] = rate;
}
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 / 8;
this->width[nr] = width;
return GST_PAD_LINK_OK;
}
@ -450,92 +454,6 @@ gst_audio_convert_change_state (GstElement *element)
}
}
/*** ACTUAL WORK **************************************************************/
#if 0
static GstCaps *
make_caps (gint endianness, gboolean sign, gint depth, gint width, gint rate, gint channels)
{
GstCaps *caps;
caps = gst_caps_new_simple ("audio/x-raw-int",
"signed", G_TYPE_BOOLEAN, sign,
"depth", G_TYPE_INT, depth,
"width", G_TYPE_INT, width * 8,
"rate", G_TYPE_INT, rate,
"channels", G_TYPE_INT, channels, NULL);
if (width != 1) {
gst_caps_set_simple (caps,
"endianness", G_TYPE_INT, endianness, NULL);
}
return caps;
}
static gboolean
gst_audio_convert_set_caps (GstPad *pad)
{
GstCaps *caps;
gint nr;
GstPadLinkReturn ret;
GstAudioConvert *this;
gint channels, endianness, depth, width; /*, rate; */
gboolean sign;
this = GST_AUDIO_CONVERT (GST_PAD_PARENT (pad));
nr = (this->src == pad) ? 1 : (this->sink == pad) ? 0 : -1;
g_assert (nr > -1);
/* try 1:1 first */
caps = make_caps (this->endian[1 - nr], this->sign[1 - nr], this->depth[1 - nr],
this->width[1 - nr], this->rate[1 - nr], this->channels[1 - nr]);
ret = gst_pad_try_set_caps (pad, caps);
if (ret == GST_PAD_LINK_DONE || ret == GST_PAD_LINK_OK) goto success;
/* now do some iterating, this is gonna be fun */
/* stereo is most important */
channels = 2;
while (channels > 0) {
/* endianness comes second */
endianness = 0;
do {
if (endianness == G_BIG_ENDIAN) break;
endianness = endianness == 0 ? G_LITTLE_ENDIAN : G_BIG_ENDIAN;
/* signedness */
sign = TRUE;
do {
sign = !sign;
/* width */
for (width = 4; width >= 1; width--) {
/* depth */
for (depth = width * 8; depth >= 1; depth -= this->aggressive ? 1 : 8) {
/* rate - not supported yet*/
caps = make_caps (endianness, sign, depth, width, this->rate[1 - nr], channels);
ret = gst_pad_try_set_caps (pad, caps);
if (ret == GST_PAD_LINK_DONE || ret == GST_PAD_LINK_OK)
goto success;
}
}
} while (sign != TRUE);
} while TRUE;
channels--;
}
return FALSE;
success:
g_assert (gst_audio_convert_link (pad, caps) == GST_PAD_LINK_OK);
return TRUE;
}
#endif
/* return a writable buffer of size which ideally is the same as before
- You must unref the new buffer
- The size of the old buffer is undefined after this operation */
@ -583,32 +501,32 @@ gst_audio_convert_buffer_to_default_format (GstAudioConvert *this, GstBuffer *bu
gint32 *dest;
guint8 *src;
if (this->width[0] == 4 && this->depth[0] == 32 &&
if (this->width[0] == 32 && this->depth[0] == 32 &&
this->endian[0] == G_BYTE_ORDER && this->sign[0] == TRUE)
return buf;
ret = gst_audio_convert_get_buffer (buf, buf->size * 4 / this->width[0]);
ret = gst_audio_convert_get_buffer (buf, buf->size * 32 / this->width[0]);
count = ret->size / 4;
src = buf->data + (count - 1) * this->width[0];
src = buf->data + (count - 1) * (this->width[0] / 8);
dest = (gint32 *) ret->data;
for (i = count - 1; i >= 0; i--) {
switch (this->width[0]) {
case 1:
case 8:
if (this->sign[0]) {
CONVERT_TO (cur, src, gint8, this->sign[0], this->endian[0], GINT8_IDENTITY, GINT8_IDENTITY);
} else {
CONVERT_TO (cur, src, guint16, this->sign[0], this->endian[0], GUINT8_IDENTITY, GUINT8_IDENTITY);
}
break;
case 2:
case 16:
if (this->sign[0]) {
CONVERT_TO (cur, src, gint16, this->sign[0], this->endian[0], GINT16_FROM_LE, GINT16_FROM_BE);
} else {
CONVERT_TO (cur, src, guint16, this->sign[0], this->endian[0], GUINT16_FROM_LE, GUINT16_FROM_BE);
}
break;
case 4:
case 32:
if (this->sign[0]) {
CONVERT_TO (cur, src, gint32, this->sign[0], this->endian[0], GINT32_FROM_LE, GINT32_FROM_BE);
} else {
@ -656,36 +574,36 @@ gst_audio_convert_buffer_from_default_format (GstAudioConvert *this, GstBuffer *
guint count, i;
gint32 *src;
if (this->width[1] == 4 && this->depth[1] == 32 &&
if (this->width[1] == 32 && this->depth[1] == 32 &&
this->endian[1] == G_BYTE_ORDER && this->sign[1] == TRUE)
return buf;
ret = gst_audio_convert_get_buffer (buf, buf->size * this->width[1] / 4);
ret = gst_audio_convert_get_buffer (buf, buf->size * this->width[1] / 32);
dest = ret->data;
src = (gint32 *) buf->data;
count = ret->size / this->width[1];
count = ret->size / (this->width[1] / 8);
for (i = 0; i < count; i++) {
gint32 int_value = *src;
src++;
switch (this->width[1]) {
case 1:
case 8:
if (this->sign[1]) {
POPULATE (gint8, GINT8_IDENTITY, GINT8_IDENTITY);
} else {
POPULATE (guint8, GUINT8_IDENTITY, GUINT8_IDENTITY);
}
break;
case 2:
case 16:
if (this->sign[1]) {
POPULATE (gint16, GINT16_TO_BE, GINT16_TO_LE);
} else {
POPULATE (guint16, GUINT16_TO_BE, GUINT16_TO_LE);
}
break;
case 4:
case 32:
if (this->sign[1]) {
POPULATE (gint32, GINT32_TO_BE, GINT32_TO_LE);
} else {
@ -741,7 +659,8 @@ plugin_init (GstPlugin *plugin)
{
if (!gst_element_register (plugin, "audioconvert", GST_RANK_NONE, GST_TYPE_AUDIO_CONVERT))
return FALSE;
if (!gst_plugin_load ("gstaudio")) return FALSE;
return TRUE;
}