mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-28 03:00:35 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/audioconvert/gstaudioconvert.c
Benjamin Otte 355318b59d gst/audioconvert/gstaudioconvert.c: set rank to PRIMARY 
Original commit message from CVS:
2004-02-22  Benjamin Otte  <otte@gnome.org>

reported by: Stefan Kost <kost@imn.htwk-leipzig.de>

* gst/audioconvert/gstaudioconvert.c: (plugin_init):
set rank to PRIMARY
* gst/volume/gstvolume.c: (plugin_init):
set rank to NONE
fixes #134960

2004-02-22   Julio M. Merino Vidal <jmmv@menta.net>

reviewed by Benjamin Otte  <otte@gnome.org>

* ext/flac/gstflacenc.c: (gst_flacenc_chain):
escape NULL strings in GST_ELEMENT_ERROR properly (fixes #135116)
2004-02-22 15:31:30 +00:00

803 lines
26 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) 2003 Benjamin Otte <in7y118@public.uni-hamburg.de>
*
* gstaudioconvert.c: Convert audio to different audio formats automatically
*
* 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.
*/
/* Element-Checklist-Version: 5 */
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <gst/gst.h>
#include <gst/floatcast/floatcast.h>
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (audio_convert_debug);
#define GST_CAT_DEFAULT (audio_convert_debug)
/*** DEFINITIONS **************************************************************/
#define GST_TYPE_AUDIO_CONVERT (gst_audio_convert_get_type())
#define GST_AUDIO_CONVERT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CONVERT,GstAudioConvert))
#define GST_AUDIO_CONVERT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_CONVERT,GstAudioConvert))
#define GST_IS_AUDIO_CONVERT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CONVERT))
#define GST_IS_AUDIO_CONVERT_CLASS(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_CONVERT))
typedef struct _GstAudioConvert GstAudioConvert;
typedef struct _GstAudioConvertCaps GstAudioConvertCaps;
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;
/* int audio caps */
gboolean sign;
gint depth;
/* float audio caps */
gint buffer_frames;
};
struct _GstAudioConvert {
GstElement element;
/* pads */
GstPad * sink;
GstPad * src;
GstAudioConvertCaps srccaps;
GstAudioConvertCaps sinkcaps;
/* conversion functions */
GstBuffer * (* convert_internal) (GstAudioConvert *this, GstBuffer *buf);
/* for int2float */
GstBuffer * output;
gint output_samples_needed;
};
struct _GstAudioConvertClass {
GstElementClass parent_class;
};
static GstElementDetails audio_convert_details = {
"Audio Conversion",
"Filter/Converter/Audio",
"Convert audio to different formats",
"Benjamin Otte <in7y118@public.uni-hamburg.de>",
};
/* type functions */
static GType gst_audio_convert_get_type (void);
static void gst_audio_convert_base_init (gpointer g_class);
static void gst_audio_convert_class_init (GstAudioConvertClass *klass);
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);
/* actual work */
#if 0
static gboolean gst_audio_convert_set_caps (GstPad *pad);
#endif
static GstBuffer * gst_audio_convert_buffer_to_default_format (GstAudioConvert *this, GstBuffer *buf);
static GstBuffer * gst_audio_convert_buffer_from_default_format (GstAudioConvert *this, GstBuffer *buf);
static GstBuffer * gst_audio_convert_channels (GstAudioConvert *this, GstBuffer *buf);
/* AudioConvert signals and args */
enum {
/* FILL ME */
LAST_SIGNAL
};
enum {
ARG_0,
ARG_AGGRESSIVE,
};
#define DEBUG_INIT(bla) \
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 *****************************************************/
static GstStaticPadTemplate gst_audio_convert_src_template =
GST_STATIC_PAD_TEMPLATE (
"src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_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 }; "
"audio/x-raw-float, " \
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) 32, "
"buffer-frames = (int) [ 0, MAX ]"
)
);
static GstStaticPadTemplate gst_audio_convert_sink_template =
GST_STATIC_PAD_TEMPLATE (
"sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_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 }; "
"audio/x-raw-float, "
"rate = (int) [ 1, MAX ],"
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) 32, "
"buffer-frames = (int) [ 0, MAX ]"
)
);
/*** TYPE FUNCTIONS ***********************************************************/
static void
gst_audio_convert_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_audio_convert_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_audio_convert_sink_template));
gst_element_class_set_details (element_class, &audio_convert_details);
}
static void
gst_audio_convert_class_init (GstAudioConvertClass *klass)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
gstelement_class->change_state = gst_audio_convert_change_state;
}
static void
gst_audio_convert_init (GstAudioConvert *this)
{
/* sinkpad */
this->sink = gst_pad_new_from_template (
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_link_function (this->sink, gst_audio_convert_link);
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_getcaps_function (this->src, gst_audio_convert_getcaps);
gst_pad_set_link_function (this->src, gst_audio_convert_link);
gst_element_add_pad (GST_ELEMENT(this), this->src);
gst_pad_set_chain_function(this->sink, gst_audio_convert_chain);
/* clear important variables */
this->convert_internal = NULL;
}
/*** GSTREAMER FUNCTIONS ******************************************************/
static void
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));
/* FIXME */
if (GST_IS_EVENT (buf)) {
gst_pad_event_default (pad, GST_EVENT (buf));
return;
}
if (!gst_pad_is_negotiated (this->sink))
{
GST_ELEMENT_ERROR (this, CORE, NEGOTIATION, NULL,
("Sink pad not negotiated before chain function"));
return;
}
if (!gst_pad_is_negotiated (this->src)) {
gst_data_unref (data);
return;
}
/**
* Theory of operation:
* - convert the format (endianness, signedness, width, depth) to
* (G_BYTE_ORDER, TRUE, 32, 32)
* - convert rate and channels
* - convert back to output format
*/
buf = gst_audio_convert_buffer_to_default_format (this, buf);
buf = gst_audio_convert_channels (this, buf);
buf = gst_audio_convert_buffer_from_default_format (this, buf);
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;
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);
g_return_val_if_fail(GST_IS_AUDIO_CONVERT(GST_OBJECT_PARENT (pad)), NULL);
this = GST_AUDIO_CONVERT(GST_OBJECT_PARENT (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++) {
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");
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, 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;
}
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;
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_int (structure, "width", &caps->width) ||
!gst_structure_get_int (structure, "rate", &caps->rate) ||
(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;
}
return TRUE;
}
static GstPadLinkReturn
gst_audio_convert_link (GstPad *pad, const GstCaps *caps)
{
GstAudioConvert *this;
GstPad *otherpad;
GstAudioConvertCaps ac_caps, other_ac_caps;
GstCaps *othercaps;
guint i;
GstPadLinkReturn ret;
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);
this = GST_AUDIO_CONVERT(GST_OBJECT_PARENT (pad));
otherpad = (pad == this->src ? this->sink : this->src);
/* negotiate sinkpad first */
if (pad == this->src &&
!gst_pad_is_negotiated (this->sink))
return GST_PAD_LINK_DELAYED;
if (!gst_audio_convert_parse_caps (caps, &ac_caps))
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;
}
/* ok, not those - try setting "any" caps */
othercaps = gst_pad_get_allowed_caps (otherpad);
for (i = 0; i < gst_caps_get_size (othercaps); i++) {
GstStructure *structure = gst_caps_get_structure (othercaps, i);
gst_structure_set (structure, "rate", G_TYPE_INT, ac_caps.rate, NULL);
}
ret = gst_pad_try_set_caps_nonfixed (otherpad, othercaps);
gst_caps_free (othercaps);
if (ret < GST_PAD_LINK_OK)
return ret;
if (!gst_audio_convert_parse_caps (caps, &other_ac_caps))
return GST_PAD_LINK_REFUSED;
/* woohoo, got it */
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;
} else {
this->srccaps = ac_caps;
this->sinkcaps = other_ac_caps;
}
return GST_PAD_LINK_OK;
}
static GstElementStateReturn
gst_audio_convert_change_state (GstElement *element)
{
GstAudioConvert *this = GST_AUDIO_CONVERT (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;
}
if (parent_class->change_state) {
return parent_class->change_state (element);
} else {
return GST_STATE_SUCCESS;
}
}
/* 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 */
static GstBuffer*
gst_audio_convert_get_buffer (GstBuffer *buf, guint size)
{
GstBuffer *ret;
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);
buf->size = size;
GST_LOG ("returning same buffer with adjusted values. data: %p - size: %u - maxsize: %u\n",
buf->data, buf->size, buf->maxsize);
return buf;
} else {
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);
return ret;
}
}
static inline guint8 GUINT8_IDENTITY (guint8 x) { return x; }
static inline guint8 GINT8_IDENTITY (gint8 x) { return x; }
#define CONVERT_TO(to, from, type, sign, endianness, LE_FUNC, BE_FUNC) G_STMT_START{\
type value; \
memcpy (&value, from, sizeof (type)); \
from -= sizeof (type); \
value = (endianness == G_LITTLE_ENDIAN) ? LE_FUNC (value) : BE_FUNC (value); \
if (sign) { \
to = value; \
} else { \
to = (gint64) value - (1 << (sizeof (type) * 8 - 1)); \
} \
}G_STMT_END;
static GstBuffer*
gst_audio_convert_buffer_to_default_format (GstAudioConvert *this, GstBuffer *buf)
{
GstBuffer *ret;
gint i, count;
gint64 cur = 0;
gint32 write;
gint32 *dest;
guint8 *src;
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);
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);
}
} 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);
return ret;
}
#define POPULATE(format, be_func, le_func) G_STMT_START{ \
format val; \
format* p = (format *) dest; \
int_value >>= (32 - this->srccaps.depth); \
val = (format) int_value; \
switch (this->srccaps.endianness) { \
case G_LITTLE_ENDIAN: \
val = le_func (val); \
break; \
case G_BIG_ENDIAN: \
val = be_func (val); \
break; \
default: \
g_assert_not_reached (); \
}; \
*p = val; \
p ++; \
dest = (guint8 *) p; \
}G_STMT_END
static GstBuffer *
gst_audio_convert_buffer_from_default_format (GstAudioConvert *this, GstBuffer *buf)
{
GstBuffer *ret;
guint8 *dest;
guint count, i;
gint32 *src;
if (this->srccaps.width == 32 && this->srccaps.depth == 32 &&
this->srccaps.endianness == G_BYTE_ORDER && this->srccaps.sign == TRUE)
return buf;
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;
src = (gint32 *) buf->data;
for (i = 0; i < count; i++) {
gint32 int_value = *src;
src++;
switch (this->srccaps.width) {
case 8:
if (this->srccaps.sign) {
POPULATE (gint8, GINT8_IDENTITY, GINT8_IDENTITY);
} else {
POPULATE (guint8, GUINT8_IDENTITY, GUINT8_IDENTITY);
}
break;
case 16:
if (this->srccaps.sign) {
POPULATE (gint16, GINT16_TO_BE, GINT16_TO_LE);
} else {
POPULATE (guint16, GUINT16_TO_BE, GUINT16_TO_LE);
}
break;
case 32:
if (this->srccaps.sign) {
POPULATE (gint32, GINT32_TO_BE, GINT32_TO_LE);
} else {
POPULATE (guint32, GUINT32_TO_BE, GUINT32_TO_LE);
}
break;
default:
g_assert_not_reached ();
}
}
gst_buffer_unref(buf);
return ret;
}
static GstBuffer *
gst_audio_convert_channels (GstAudioConvert *this, GstBuffer *buf)
{
GstBuffer *ret;
gint i, count;
guint32 *src, *dest;
if (this->sinkcaps.channels == this->srccaps.channels)
return buf;
count = GST_BUFFER_SIZE (buf) / 4 / this->sinkcaps.channels;
ret = gst_audio_convert_get_buffer (buf, count * 4 * this->srccaps.channels);
src = (guint32 *) GST_BUFFER_DATA (buf);
dest = (guint32 *) GST_BUFFER_DATA (ret);
if (this->sinkcaps.channels > this->srccaps.channels) {
for (i = 0; i < count; i++) {
*dest = *src >> 1;
src++;
*dest += (*src + 1) >> 1;
src++;
dest++;
}
} else {
for (i = count - 1; i >= 0; i--) {
dest[2 * i] = dest[2 * i + 1] = src[i];
}
}
gst_buffer_unref(buf);
return ret;
}
/*** PLUGIN DETAILS ***********************************************************/
static gboolean
plugin_init (GstPlugin *plugin)
{
if (!gst_element_register (plugin, "audioconvert", GST_RANK_PRIMARY, GST_TYPE_AUDIO_CONVERT))
return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (
GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"gstaudioconvert",
"Convert audio to different formats",
plugin_init,
VERSION,
"LGPL",
GST_PACKAGE,
GST_ORIGIN)
Reference in a new issue View git blame Copy permalink