mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-16 21:36:35 +00:00
Code Issues 1 Projects Releases Wiki Activity

gst/intfloat/, gst/oneton: Removed, replaced by audioconvert and interleave respectively.

Browse source 
Original commit message from CVS:
2004-02-20  Andy Wingo  <wingo@pobox.com>

* gst/intfloat/, gst/oneton: Removed, replaced by audioconvert and
interleave respectively.

* gst/interleave/deinterleave.c: New plugin: deinterleave
(replaces on oneton).
* gst/interleave/interleave.c: New plugin: interleave.
* gst/interleave/plugin.h: Support file.
* gst/interleave/plugin.c: Support file.

* configure.ac: Remove intfloat and oneton, add interleave.

* ext/sndfile/gstsf.c: Handle events better.

* gst/audioconvert/gstaudioconvert.c: Change to support int2float
and float2int operation. int2float has scheduling problems as
noted in in2float_chain.
This commit is contained in:
Andy Wingo 2004-02-20 14:17:57 +00:00
parent 8eae64c32d
commit ce89f16818
3 changed files with 281 additions and 60 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

19
ChangeLog
View file

@ -1,3 +1,22 @@
2004-02-20 Andy Wingo <wingo@pobox.com>
* gst/intfloat/, gst/oneton: Removed, replaced by audioconvert and
interleave respectively.
* gst/interleave/deinterleave.c: New plugin: deinterleave
(replaces on oneton).
* gst/interleave/interleave.c: New plugin: interleave.
* gst/interleave/plugin.h: Support file.
* gst/interleave/plugin.c: Support file.
* configure.ac: Remove intfloat and oneton, add interleave.
* ext/sndfile/gstsf.c: Handle events better.
* gst/audioconvert/gstaudioconvert.c: Change to support int2float
and float2int operation. int2float has scheduling problems as
noted in in2float_chain.
2004-02-20 Benjamin Otte <otte@gnome.org>
* ext/xine/Makefile.am:

6
configure.ac
View file

@ -346,7 +346,7 @@ GST_PLUGINS_ALL="\
flx \
goom \
id3 \
intfloat \
interleave \
law \
level \
matroska \
@ -359,7 +359,6 @@ GST_PLUGINS_ALL="\
mpegaudioparse \
mpegstream \
monoscope \
oneton \
overlay \
passthrough \
playondemand \
@ -1559,7 +1558,7 @@ gst/filter/Makefile
gst/flx/Makefile
gst/goom/Makefile
gst/id3/Makefile
gst/intfloat/Makefile
gst/interleave/Makefile
gst/law/Makefile
gst/level/Makefile
gst/matroska/Makefile
@ -1574,7 +1573,6 @@ gst/mpegstream/Makefile
gst/modplug/Makefile
gst/modplug/libmodplug/Makefile
gst/monoscope/Makefile
gst/oneton/Makefile
gst/overlay/Makefile
gst/passthrough/Makefile
gst/playondemand/Makefile

316
gst/audioconvert/gstaudioconvert.c
View file

@ -25,6 +25,7 @@
#endif
#include <gst/gst.h>
#include <gst/floatcast/floatcast.h>
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (audio_convert_debug);
@ -45,14 +46,18 @@ typedef struct _GstAudioConvertClass GstAudioConvertClass;
/* this struct is a handy way of passing around all the caps info ... */
struct _GstAudioConvertCaps {
/* general caps */
gboolean is_int;
gint endianness;
gint width;
gint rate;
gint channels;
gboolean sign;
/* int audio caps */
gboolean sign;
gint depth;
/* float audio caps */
gint buffer_frames;
};
struct _GstAudioConvert {
@ -67,6 +72,10 @@ struct _GstAudioConvert {
/* conversion functions */
GstBuffer * (* convert_internal) (GstAudioConvert *this, GstBuffer *buf);
/* for int2float */
GstBuffer * output;
gint output_samples_needed;
};
struct _GstAudioConvertClass {
@ -77,7 +86,7 @@ static GstElementDetails audio_convert_details = {
"Audio Conversion",
"Filter/Converter/Audio",
"Convert audio to different formats",
"Benjamin Otte <in7y118@public.uni-hamburg.de",
"Benjamin Otte <in7y118@public.uni-hamburg.de>",
};
/* type functions */
@ -88,6 +97,7 @@ static void gst_audio_convert_init (GstAudioConvert *audio_convert);
/* gstreamer functions */
static void gst_audio_convert_chain (GstPad *pad, GstData *_data);
static void gst_audio_convert_chain_int2float (GstPad *pad, GstData *_data);
static GstPadLinkReturn gst_audio_convert_link (GstPad *pad, const GstCaps *caps);
static GstCaps * gst_audio_convert_getcaps (GstPad *pad);
static GstElementStateReturn gst_audio_convert_change_state (GstElement *element);
@ -132,7 +142,14 @@ GST_STATIC_PAD_TEMPLATE (
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) { 8, 16, 32 }, " \
"depth = (int) [ 1, 32 ], " \
"signed = (boolean) { true, false }"
"signed = (boolean) { true, false }; "
"audio/x-raw-float, " \
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) 32, "
"buffer-frames = (int) [ 0, MAX ]"
)
);
@ -148,7 +165,14 @@ GST_STATIC_PAD_TEMPLATE (
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) { 8, 16, 32 }, " \
"depth = (int) [ 1, 32 ], " \
"signed = (boolean) { true, false }"
"signed = (boolean) { true, false }; "
"audio/x-raw-float, "
"rate = (int) [ 1, MAX ],"
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) 32, "
"buffer-frames = (int) [ 0, MAX ]"
)
);
@ -244,13 +268,135 @@ gst_audio_convert_chain (GstPad *pad, GstData *data)
gst_pad_push (this->src, GST_DATA (buf));
}
/* 1 / (2^31-1) * i */
#define INT2FLOAT(i) (4.6566128752457969e-10 * ((gfloat)i))
/* This custom chain handler exists because if buffer-frames is nonzero, one int
* buffer probably doesn't correspond to one float buffer */
static void
gst_audio_convert_chain_int2float (GstPad *pad, GstData *data)
{
GstBuffer *buf = GST_BUFFER (data);
GstAudioConvert *this;
gint buffer_samples, samples_remaining, i;
gint32 *in;
gfloat *out;
this = GST_AUDIO_CONVERT (GST_OBJECT_PARENT (pad));
/* FIXME */
if (GST_IS_EVENT (buf)) {
gst_pad_event_default (pad, GST_EVENT (buf));
return;
}
/* we know we're negotiated, because it's the link function that set the
custom chain handler */
/* FIXME: this runs into scheduling problems if the next element is loop-based
* (the bufpen fills up until infinity because we push multiple buffers per
* chain, in the normal situation). The fix is either to make the opt
* scheduler choose the loop group as its entry, or to make this a loop
* plugin. But I want to commit, will fix this later. */
/**
* Theory of operation:
* - convert the format (endianness, signedness, width, depth) to
* (G_BYTE_ORDER, TRUE, 32, 32)
* - convert rate and channels
* - if buffer-frames is zero, convert and push.
* - if we have an output buffer, fill it. if it becomes full, push it.
* - while buffer-frames is less than the number of frames remaining in the
* input, create sub-buffers, convert and push.
* - if there are leftover frames in the input, create an output buffer and
* fill it partially.
*/
buf = gst_audio_convert_buffer_to_default_format (this, buf);
buf = gst_audio_convert_channels (this, buf);
/* we know buf is writable */
buffer_samples = this->srccaps.buffer_frames * this->srccaps.channels;
in = (gint32*)GST_BUFFER_DATA (buf);
out = (gfloat*)GST_BUFFER_DATA (buf);
samples_remaining = buf->size / sizeof(gint32);
if (!buffer_samples ||
(!this->output && samples_remaining == buffer_samples)) {
for (i=samples_remaining; i; i--)
*(out++) = INT2FLOAT (*(in++));
gst_pad_push (this->src, GST_DATA (buf));
return;
}
if (this->output) {
GstBuffer *output = this->output;
gint to_process = MIN (this->output_samples_needed, samples_remaining);
out = ((gfloat*)GST_BUFFER_DATA (output) +
(buffer_samples - this->output_samples_needed));
for (i=to_process; i; i--)
*(out++) = INT2FLOAT (*(in++));
this->output_samples_needed -= to_process;
samples_remaining -= to_process;
/* one of the two of these ifs will be true, and possibly both of them */
if (!this->output_samples_needed) {
this->output = NULL;
gst_pad_push (this->src, GST_DATA (output));
}
if (!samples_remaining) {
gst_buffer_unref (buf);
return;
}
/* we have some leftover frames in buf, let's take care of them */
out = (gfloat*)in;
}
while (samples_remaining > buffer_samples) {
GstBuffer *sub_buf;
sub_buf = gst_buffer_create_sub (buf,
(GST_BUFFER_SIZE (buf) -
samples_remaining * sizeof(gint32)),
buffer_samples * sizeof(gfloat));
/* `out' should be positioned correctly */
for (i=buffer_samples; i; i--)
*(out++) = INT2FLOAT (*(in++));
samples_remaining -= buffer_samples;
gst_pad_push (this->src, GST_DATA (sub_buf));
}
if (samples_remaining) {
GstBuffer *output;
output = this->output = gst_buffer_new_and_alloc (buffer_samples * sizeof(gfloat));
out = (gfloat*)GST_BUFFER_DATA (output);
for (i=samples_remaining; i; i--)
*(out++) = INT2FLOAT (*(in++));
this->output = output;
this->output_samples_needed = buffer_samples - samples_remaining;
samples_remaining = 0; /* just so we know */
}
gst_buffer_unref (buf);
return;
}
/* this function is complicated now, but it will be unnecessary when we convert
* rate. */
static GstCaps *
gst_audio_convert_getcaps (GstPad *pad)
{
GstAudioConvert *this;
GstPad *otherpad;
GstCaps *othercaps;
GstCaps *caps;
GstStructure *structure;
GstCaps *othercaps, *caps;
const GstCaps *templcaps;
gboolean has_float = FALSE, has_int = FALSE;
int i;
g_return_val_if_fail(GST_IS_PAD(pad), NULL);
@ -259,21 +405,44 @@ gst_audio_convert_getcaps (GstPad *pad)
otherpad = (pad == this->src) ? this->sink : this->src;
/* all we want to find out is the rate */
templcaps = gst_pad_get_pad_template_caps (pad);
othercaps = gst_pad_get_allowed_caps (otherpad);
for (i=0;i<gst_caps_get_size (othercaps); i++) {
GstStructure *structure;
structure = gst_caps_get_structure (othercaps, i);
gst_structure_remove_field (structure, "channels");
gst_structure_remove_field (structure, "endianness");
gst_structure_remove_field (structure, "width");
gst_structure_remove_field (structure, "depth");
gst_structure_remove_field (structure, "signed");
if (strcmp (gst_structure_get_name (structure), "audio/x-raw-int") == 0) {
if (!has_int) has_int = TRUE;
gst_structure_remove_field (structure, "depth");
gst_structure_remove_field (structure, "signed");
} else {
if (!has_float) has_float = TRUE;
gst_structure_remove_field (structure, "buffer-frames");
}
}
caps = gst_caps_intersect (othercaps, gst_pad_get_pad_template_caps (pad));
caps = gst_caps_intersect (othercaps, templcaps);
gst_caps_free (othercaps);
/* the intersection probably lost either float or int. so we take the rate
* property and set it on a copy of the templcaps struct. */
if (!has_int) {
structure = gst_structure_copy (gst_caps_get_structure (templcaps, 0));
gst_structure_set_value (structure, "rate",
gst_structure_get_value (gst_caps_get_structure (caps, 0),
"rate"));
gst_caps_append_structure (caps, structure);
}
if (!has_float) {
structure = gst_structure_copy (gst_caps_get_structure (templcaps, 1));
gst_structure_set_value (structure, "rate",
gst_structure_get_value (gst_caps_get_structure (caps, 0),
"rate"));
gst_caps_append_structure (caps, structure);
}
return caps;
}
@ -285,13 +454,17 @@ gst_audio_convert_parse_caps (const GstCaps* gst_caps, GstAudioConvertCaps *caps
g_return_val_if_fail (caps != NULL, FALSE);
caps->endianness = G_BYTE_ORDER;
caps->is_int = (strcmp (gst_structure_get_name (structure), "audio/x-raw-int") == 0);
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))) {
(caps->is_int &&
(!gst_structure_get_boolean (structure, "signed", &caps->sign) ||
!gst_structure_get_int (structure, "depth", &caps->depth) ||
(caps->width != 8 &&
!gst_structure_get_int (structure, "endianness", &caps->endianness)))) ||
(!caps->is_int &&
!gst_structure_get_int (structure, "buffer-frames", &caps->buffer_frames))) {
GST_DEBUG ("could not get some values from structure");
return FALSE;
}
@ -343,9 +516,21 @@ gst_audio_convert_link (GstPad *pad, const GstCaps *caps)
return GST_PAD_LINK_REFUSED;
/* woohoo, got it */
g_return_val_if_fail (gst_audio_convert_parse_caps (
gst_pad_get_negotiated_caps (otherpad), &other_ac_caps),
GST_PAD_LINK_REFUSED);
if (!gst_audio_convert_parse_caps (gst_pad_get_negotiated_caps (otherpad),
&other_ac_caps)) {
g_critical ("internal negotiation error");
return GST_PAD_LINK_REFUSED;
}
if (!other_ac_caps.is_int && !ac_caps.is_int) {
GST_DEBUG ("we don't do float-float conversions yet");
return GST_PAD_LINK_REFUSED;
} else if ((this->sink == pad) ? !other_ac_caps.is_int : ac_caps.is_int) {
GST_DEBUG ("int-float conversion, setting custom chain handler");
gst_pad_set_chain_function (this->sink, gst_audio_convert_chain_int2float);
}
/* float2int conversion is handled like other int formats */
if (this->sink == pad) {
this->srccaps = other_ac_caps;
this->sinkcaps = ac_caps;
@ -365,6 +550,8 @@ gst_audio_convert_change_state (GstElement *element)
switch (GST_STATE_TRANSITION (element)) {
case GST_STATE_PAUSED_TO_READY:
this->convert_internal = NULL;
GST_DEBUG ("resetting chain function to the default");
gst_pad_set_chain_function (this->sink, gst_audio_convert_chain);
break;
default:
break;
@ -384,7 +571,7 @@ static GstBuffer*
gst_audio_convert_get_buffer (GstBuffer *buf, guint size)
{
GstBuffer *ret;
GST_LOG ("new buffer of size %u requersted. Current is: data: %p - size: %u - maxsize: %u\n",
GST_LOG ("new buffer of size %u requested. 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)) {
gst_buffer_ref (buf);
@ -393,7 +580,8 @@ gst_audio_convert_get_buffer (GstBuffer *buf, guint size)
buf->data, buf->size, buf->maxsize);
return buf;
} else {
g_assert ((ret = gst_buffer_new_and_alloc (size)));
ret = gst_buffer_new_and_alloc (size);
g_assert (ret);
gst_buffer_stamp (ret, buf);
GST_LOG ("returning new buffer. data: %p - size: %u - maxsize: %u\n",
ret->data, ret->size, ret->maxsize);
@ -426,45 +614,61 @@ gst_audio_convert_buffer_to_default_format (GstAudioConvert *this, GstBuffer *bu
gint32 *dest;
guint8 *src;
if (this->sinkcaps.width == 32 && this->sinkcaps.depth == 32 &&
this->sinkcaps.endianness == G_BYTE_ORDER && this->sinkcaps.sign == TRUE)
return buf;
if (this->sinkcaps.is_int) {
if (this->sinkcaps.width == 32 && this->sinkcaps.depth == 32 &&
this->sinkcaps.endianness == G_BYTE_ORDER && this->sinkcaps.sign == TRUE)
return buf;
ret = gst_audio_convert_get_buffer (buf, buf->size * 32 / this->sinkcaps.width);
ret = gst_audio_convert_get_buffer (buf, buf->size * 32 / this->sinkcaps.width);
count = ret->size / 4;
src = buf->data + (count - 1) * (this->sinkcaps.width / 8);
dest = (gint32 *) ret->data;
for (i = count - 1; i >= 0; i--) {
switch (this->sinkcaps.width) {
case 8:
if (this->sinkcaps.sign) {
CONVERT_TO (cur, src, gint8, this->sinkcaps.sign, this->sinkcaps.endianness, GINT8_IDENTITY, GINT8_IDENTITY);
} else {
CONVERT_TO (cur, src, guint16, this->sinkcaps.sign, this->sinkcaps.endianness, GUINT8_IDENTITY, GUINT8_IDENTITY);
count = ret->size / 4;
src = buf->data + (count - 1) * (this->sinkcaps.width / 8);
dest = (gint32 *) ret->data;
for (i = count - 1; i >= 0; i--) {
switch (this->sinkcaps.width) {
case 8:
if (this->sinkcaps.sign) {
CONVERT_TO (cur, src, gint8, this->sinkcaps.sign, this->sinkcaps.endianness, GINT8_IDENTITY, GINT8_IDENTITY);
} else {
CONVERT_TO (cur, src, guint16, this->sinkcaps.sign, this->sinkcaps.endianness, GUINT8_IDENTITY, GUINT8_IDENTITY);
}
break;
case 16:
if (this->sinkcaps.sign) {
CONVERT_TO (cur, src, gint16, this->sinkcaps.sign, this->sinkcaps.endianness, GINT16_FROM_LE, GINT16_FROM_BE);
} else {
CONVERT_TO (cur, src, guint16, this->sinkcaps.sign, this->sinkcaps.endianness, GUINT16_FROM_LE, GUINT16_FROM_BE);
}
break;
case 32:
if (this->sinkcaps.sign) {
CONVERT_TO (cur, src, gint32, this->sinkcaps.sign, this->sinkcaps.endianness, GINT32_FROM_LE, GINT32_FROM_BE);
} else {
CONVERT_TO (cur, src, guint32, this->sinkcaps.sign, this->sinkcaps.endianness, GUINT32_FROM_LE, GUINT32_FROM_BE);
}
break;
default:
g_assert_not_reached ();
}
cur = cur * ((gint64) 1 << (32 - this->sinkcaps.depth));
cur = CLAMP (cur, -((gint64)1 << 32), (gint64) 0x7FFFFFFF);
write = cur;
memcpy (&dest[i], &write, 4);
}
break;
case 16:
if (this->sinkcaps.sign) {
CONVERT_TO (cur, src, gint16, this->sinkcaps.sign, this->sinkcaps.endianness, GINT16_FROM_LE, GINT16_FROM_BE);
} else {
CONVERT_TO (cur, src, guint16, this->sinkcaps.sign, this->sinkcaps.endianness, GUINT16_FROM_LE, GUINT16_FROM_BE);
}
break;
case 32:
if (this->sinkcaps.sign) {
CONVERT_TO (cur, src, gint32, this->sinkcaps.sign, this->sinkcaps.endianness, GINT32_FROM_LE, GINT32_FROM_BE);
} else {
CONVERT_TO (cur, src, guint32, this->sinkcaps.sign, this->sinkcaps.endianness, GUINT32_FROM_LE, GUINT32_FROM_BE);
}
break;
default:
g_assert_not_reached ();
}
cur = cur * ((gint64) 1 << (32 - this->sinkcaps.depth));
cur = CLAMP (cur, -((gint64)1 << 32), (gint64) 0x7FFFFFFF);
write = cur;
memcpy (&dest[i], &write, 4);
} else {
/* float2int */
gfloat *in;
gint32 *out;
/* should just give the same buffer, unless it's not writable -- float is
* already 32 bits */
ret = gst_audio_convert_get_buffer (buf, buf->size);
in = (gfloat*)GST_BUFFER_DATA (buf);
out = (gint32*)GST_BUFFER_DATA (ret);
/* increment `in' via the for, cause CLAMP duplicates the first arg */
for (i = buf->size / sizeof(float); i; i--, in++)
*(out++) = (gint32) gst_cast_float(CLAMP (*in, -1.f, 1.f) * 2147483647.0F);
}
gst_buffer_unref (buf);