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gstreamer/gst/audioconvert/gstaudioconvert.c
Ronald S. Bultje 78016b40cf gst/audioconvert/gstaudioconvert.c: Implement instant setup switching. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
(gst_audio_convert_chain), (gst_audio_convert_link_src),
(gst_audio_convert_setcaps):
Implement instant setup switching.
2005-05-23 17:28:02 +00:00

967 lines
30 KiB
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/* 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 */
/*
* design decisions:
* - audioconvert converts buffers in a set of supported caps. If it supports
* a caps, it supports conversion from these caps to any other caps it
* supports. (example: if it does A=>B and A=>C, it also does B=>C)
* - audioconvert does not save state between buffers. Every incoming buffer is
* converted and the converted buffer is pushed out.
* conclusion:
* audioconvert is not supposed to be a one-element-does-anything solution for
* audio conversions.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/audio/multichannel.h>
#include <string.h>
#include "gstchannelmix.h"
#include "plugin.h"
GST_DEBUG_CATEGORY (audio_convert_debug);
/*** DEFINITIONS **************************************************************/
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 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);
static void gst_audio_convert_dispose (GObject * obj);
/* gstreamer functions */
static GstFlowReturn gst_audio_convert_chain (GstPad * pad, GstBuffer * buffer);
static gboolean gst_audio_convert_link_src (GstAudioConvert * this,
GstCaps * sinkcaps, GstAudioConvertCaps * sink_ac_caps);
static gboolean gst_audio_convert_setcaps (GstPad * pad, GstCaps * caps);
static GstCaps *gst_audio_convert_fixate (GstPad * pad, GstCaps * caps);
static GstCaps *gst_audio_convert_getcaps (GstPad * pad);
static GstElementStateReturn gst_audio_convert_change_state (GstElement *
element);
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 *****************************************************/
#define STATIC_CAPS \
GST_STATIC_CAPS ( \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 8, " \
"depth = (int) [ 1, 8 ], " \
"signed = (boolean) { true, false }; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 16, " \
"depth = (int) [ 1, 16 ], " \
"signed = (boolean) { true, false }; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 24, " \
"depth = (int) [ 1, 24 ], " \
"signed = (boolean) { true, false }; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 32, " \
"depth = (int) [ 1, 32 ], " \
"signed = (boolean) { true, false }; " \
"audio/x-raw-float, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 32, " \
"buffer-frames = (int) [ 0, MAX ]" \
)
static GstAudioChannelPosition *supported_positions;
static GstStaticPadTemplate gst_audio_convert_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
STATIC_CAPS);
static GstStaticPadTemplate gst_audio_convert_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
STATIC_CAPS);
/*** 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);
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
gint i;
gstelement_class->change_state = gst_audio_convert_change_state;
gobject_class->dispose = gst_audio_convert_dispose;
supported_positions = g_new0 (GstAudioChannelPosition,
GST_AUDIO_CHANNEL_POSITION_NUM);
for (i = 0; i < GST_AUDIO_CHANNEL_POSITION_NUM; i++)
supported_positions[i] = i;
}
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_setcaps_function (this->sink, gst_audio_convert_setcaps);
gst_pad_set_fixatecaps_function (this->sink, gst_audio_convert_fixate);
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_setcaps_function (this->src, gst_audio_convert_setcaps);
gst_pad_set_fixatecaps_function (this->src, gst_audio_convert_fixate);
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;
this->sinkcaps.pos = NULL;
this->srccaps.pos = NULL;
this->matrix = NULL;
}
static void
gst_audio_convert_dispose (GObject * obj)
{
GstAudioConvert *this = GST_AUDIO_CONVERT (obj);
if (this->sinkcaps.pos) {
g_free (this->sinkcaps.pos);
this->sinkcaps.pos = NULL;
}
if (this->srccaps.pos) {
g_free (this->srccaps.pos);
this->srccaps.pos = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (obj);
}
/*** GSTREAMER FUNCTIONS ******************************************************/
static GstFlowReturn
gst_audio_convert_chain (GstPad * pad, GstBuffer * buf)
{
GstAudioConvert *this;
GstFlowReturn ret;
this = GST_AUDIO_CONVERT (GST_OBJECT_PARENT (pad));
/**
* 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
*/
GST_STREAM_LOCK (pad);
if (!GST_RPAD_CAPS (this->sink)) {
goto not_negotiated;
} else if (!GST_RPAD_CAPS (this->src)) {
if (!gst_audio_convert_link_src (this,
GST_RPAD_CAPS (this->sink), &this->sinkcaps))
goto no_format;
} else if (!this->matrix) {
gst_audio_convert_setup_matrix (this);
}
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);
ret = gst_pad_push (this->src, buf);
GST_STREAM_UNLOCK (pad);
return ret;
not_negotiated:
{
GST_ELEMENT_ERROR (this, CORE, NEGOTIATION, (NULL),
("Pad not negotiated before chain function was called"));
gst_buffer_unref (buf);
GST_STREAM_UNLOCK (pad);
return GST_FLOW_NOT_NEGOTIATED;
}
no_format:
{
GST_ELEMENT_ERROR (this, CORE, NEGOTIATION, (NULL),
("Could not negotiate format"));
gst_buffer_unref (buf);
GST_STREAM_UNLOCK (pad);
return GST_FLOW_ERROR;
}
}
static GstCaps *
gst_audio_convert_caps_remove_format_info (GstPad * pad, GstCaps * caps)
{
int i, size;
GstAudioConvert *this;
this = GST_AUDIO_CONVERT (GST_OBJECT_PARENT (pad));
size = gst_caps_get_size (caps);
caps = gst_caps_make_writable (caps);
for (i = size - 1; i >= 0; i--) {
GstStructure *structure;
structure = gst_caps_get_structure (caps, i);
gst_structure_remove_field (structure, "channels");
gst_structure_remove_field (structure, "channel-positions");
gst_structure_remove_field (structure, "endianness");
gst_structure_remove_field (structure, "width");
gst_structure_remove_field (structure, "depth");
gst_structure_remove_field (structure, "signed");
structure = gst_structure_copy (structure);
if (strcmp (gst_structure_get_name (structure), "audio/x-raw-int") == 0) {
gst_structure_set_name (structure, "audio/x-raw-float");
if (pad == this->sink) {
gst_structure_set (structure, "buffer-frames", GST_TYPE_INT_RANGE, 0,
G_MAXINT, NULL);
} else {
gst_structure_set (structure, "buffer-frames", G_TYPE_INT, 0, NULL);
}
} else {
gst_structure_set_name (structure, "audio/x-raw-int");
gst_structure_remove_field (structure, "buffer-frames");
}
gst_caps_append_structure (caps, structure);
}
return caps;
}
/* 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, *caps;
const GstCaps *templcaps;
this = GST_AUDIO_CONVERT (GST_OBJECT_PARENT (pad));
otherpad = (pad == this->src) ? this->sink : this->src;
/* we can do all our peer can */
othercaps = gst_pad_peer_get_caps (otherpad);
if (othercaps != NULL) {
/* without the format info even */
othercaps = gst_audio_convert_caps_remove_format_info (pad, othercaps);
/* but filtered against our template */
templcaps = gst_pad_get_pad_template_caps (pad);
caps = gst_caps_intersect (othercaps, templcaps);
gst_caps_unref (othercaps);
} else {
/* no peer, then our template is enough */
caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
}
/* Get the channel positions in as well. */
gst_audio_set_caps_channel_positions_list (caps, supported_positions,
GST_AUDIO_CHANNEL_POSITION_NUM);
return caps;
}
static gboolean
gst_audio_convert_parse_caps (const GstCaps * gst_caps,
GstAudioConvertCaps * caps)
{
GstStructure *structure = gst_caps_get_structure (gst_caps, 0);
GST_DEBUG ("parse caps %p and %" GST_PTR_FORMAT, gst_caps, gst_caps);
g_return_val_if_fail (gst_caps_is_fixed (gst_caps), FALSE);
g_return_val_if_fail (caps != NULL, FALSE);
/* cleanup old */
if (caps->pos) {
g_free (caps->pos);
caps->pos = NULL;
}
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)
|| !(caps->pos = gst_audio_get_channel_positions (structure))
|| !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");
g_free (caps->pos);
caps->pos = NULL;
return FALSE;
}
if (caps->is_int && caps->depth > caps->width) {
GST_DEBUG ("width > depth, not allowed - make us advertise correct caps");
g_free (caps->pos);
caps->pos = NULL;
return FALSE;
}
return TRUE;
}
static gboolean
gst_audio_convert_link_src (GstAudioConvert * this,
GstCaps * sinkcaps, GstAudioConvertCaps * sink_ac_caps)
{
GstAudioConvertCaps ac_caps = { 0 };
if (gst_pad_peer_accept_caps (this->src, sinkcaps)) {
/* great, so that will be our suggestion then */
this->src_prefered = gst_caps_ref (sinkcaps);
gst_caps_replace (&GST_RPAD_CAPS (this->src), sinkcaps);
ac_caps = *sink_ac_caps;
if (ac_caps.pos) {
ac_caps.pos = g_memdup (ac_caps.pos, sizeof (gint) * ac_caps.channels);
}
} else {
/* nope, find something we can convert to and the peer can
* accept. */
GstCaps *othercaps = gst_pad_peer_get_caps (this->src);
if (othercaps) {
/* peel off first one */
GstCaps *targetcaps = gst_caps_copy_nth (othercaps, 0);
GstStructure *structure = gst_caps_get_structure (targetcaps, 0);
gst_caps_unref (othercaps);
/* set the rate on the caps, this has to work */
gst_structure_set (structure,
"rate", G_TYPE_INT, sink_ac_caps->rate,
"channels", G_TYPE_INT, sink_ac_caps->channels, NULL);
if (strcmp (gst_structure_get_name (structure), "audio/x-raw-float") == 0) {
if (!sink_ac_caps->is_int) {
/* copy over */
gst_structure_set (structure, "buffer-frames", G_TYPE_INT,
ac_caps.buffer_frames, NULL);
} else {
/* set to anything */
gst_structure_set (structure, "buffer-frames", G_TYPE_INT, 0, NULL);
}
}
/* this will be our suggestion */
this->src_prefered = targetcaps;
if (!gst_audio_convert_parse_caps (targetcaps, &ac_caps))
return FALSE;
gst_caps_replace (&GST_RPAD_CAPS (this->src), targetcaps);
}
}
this->srccaps = ac_caps;
GST_DEBUG_OBJECT (this, "negotiated pad to %" GST_PTR_FORMAT, sinkcaps);
return TRUE;
}
static gboolean
gst_audio_convert_setcaps (GstPad * pad, GstCaps * caps)
{
GstAudioConvert *this;
GstAudioConvertCaps ac_caps = { 0 };
gboolean res;
g_return_val_if_fail (GST_IS_PAD (pad), FALSE);
g_return_val_if_fail (GST_IS_AUDIO_CONVERT (GST_OBJECT_PARENT (pad)), FALSE);
g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
this = GST_AUDIO_CONVERT (GST_OBJECT_PARENT (pad));
/* we'll need a new matrix after every new negotiation */
gst_audio_convert_unset_matrix (this);
ac_caps.pos = NULL;
if (!gst_audio_convert_parse_caps (caps, &ac_caps))
return FALSE;
this->sink_prefered = caps;
if ((res = gst_audio_convert_link_src (this, caps, &ac_caps))) {
this->sinkcaps = ac_caps;
GST_DEBUG_OBJECT (this, "negotiated pad to %" GST_PTR_FORMAT, caps);
}
return res;
}
/* tries to fixate the given field of the given caps to the given int value */
gboolean
_fixate_caps_to_int (GstCaps ** caps, const gchar * field, gint value)
{
GstCaps *try, *isect_lower, *isect_higher;
gboolean ret = FALSE;
guint i;
/* First try to see if we can fixate by intersecting given caps with
* simple audio caps with ranges starting/ending with value */
try = gst_caps_new_simple ("audio/x-raw-int", field, GST_TYPE_INT_RANGE,
G_MININT, value - 1, NULL);
gst_caps_append (try, gst_caps_new_simple ("audio/x-raw-float", field,
GST_TYPE_INT_RANGE, G_MININT, value - 1, NULL));
isect_lower = gst_caps_intersect (*caps, try);
gst_caps_unref (try);
if (!gst_caps_is_empty (isect_lower)) {
try = gst_caps_new_simple ("audio/x-raw-int", field, GST_TYPE_INT_RANGE,
value, G_MAXINT, NULL);
gst_caps_append (try, gst_caps_new_simple ("audio/x-raw-float", field,
GST_TYPE_INT_RANGE, value, G_MAXINT, NULL));
isect_higher = gst_caps_intersect (*caps, try);
gst_caps_unref (try);
/* FIXME: why choose to end up with the higher range, and not the fixed
* value ? */
if (!gst_caps_is_empty (isect_higher)) {
gst_caps_unref (*caps);
*caps = isect_higher;
ret = TRUE;
} else {
gst_caps_unref (isect_higher);
}
}
gst_caps_unref (isect_lower);
/* FIXME: why don't we already return here when ret == TRUE ? */
for (i = 0; i < gst_caps_get_size (*caps); i++) {
GstStructure *structure = gst_caps_get_structure (*caps, i);
if (gst_structure_has_field (structure, field))
ret |=
gst_caps_structure_fixate_field_nearest_int (structure, field, value);
}
return ret;
}
static GstCaps *
gst_audio_convert_fixate (GstPad * pad, GstCaps * caps)
{
const GValue *pos_val;
GstAudioConvert *this =
GST_AUDIO_CONVERT (gst_object_get_parent (GST_OBJECT (pad)));
//GstPad *otherpad = (pad == this->sink ? this->src : this->sink);
GstAudioConvertCaps try, ac_caps =
(pad == this->sink ? this->srccaps : this->sinkcaps);
GstCaps *copy = gst_caps_copy (caps);
//if (!GST_PAD_IS_NEGOTIATING (otherpad)) {
try.channels = 2;
try.width = 16;
try.depth = 16;
try.endianness = G_BYTE_ORDER;
/*
} else {
try.channels = ac_caps.channels;
try.width = ac_caps.is_int ? ac_caps.width : 16;
try.depth = ac_caps.is_int ? ac_caps.depth : 16;
try.endianness = ac_caps.is_int ? ac_caps.endianness : G_BYTE_ORDER;
}
*/
if (_fixate_caps_to_int (&copy, "channels", try.channels)) {
int n, c;
gst_structure_get_int (gst_caps_get_structure (copy, 0), "channels", &c);
if (c > 2) {
/* make sure we have a channelpositions structure or array here */
GstStructure *str;
for (n = 0; n < gst_caps_get_size (copy); n++) {
str = gst_caps_get_structure (copy, n);
if (!gst_structure_get_value (str, "channel-positions")) {
/* first try otherpad's positions, else anything */
if (ac_caps.pos != NULL && c == ac_caps.channels) {
gst_audio_set_channel_positions (str, ac_caps.pos);
} else {
gst_audio_set_structure_channel_positions_list (str,
supported_positions, GST_AUDIO_CHANNEL_POSITION_NUM);
/* FIXME: fixate (else we'll be less fixed than we used to) */
}
}
}
} else {
/* make sure we don't */
for (n = 0; n < gst_caps_get_size (copy); n++) {
gst_structure_remove_field (gst_caps_get_structure (copy, n),
"channel-positions");
}
}
return copy;
}
if (_fixate_caps_to_int (&copy, "width", try.width))
return copy;
if (gst_structure_get_name (gst_caps_get_structure (copy, 0))[12] == 'i') {
if (_fixate_caps_to_int (&copy, "depth", try.depth))
return copy;
}
if (_fixate_caps_to_int (&copy, "endianness", try.endianness))
return copy;
if ((pos_val = gst_structure_get_value (gst_caps_get_structure (copy, 0),
"channel-positions")) != NULL) {
GstAudioChannelPosition *pos;
const GValue *pos_val_entry;
gint i;
for (i = 0; i < gst_value_list_get_size (pos_val); i++) {
pos_val_entry = gst_value_list_get_value (pos_val, i);
if (G_VALUE_TYPE (pos_val_entry) == GST_TYPE_LIST) {
/* unfixed */
pos =
gst_audio_fixate_channel_positions (gst_caps_get_structure (copy,
0));
if (pos) {
gst_audio_set_channel_positions (gst_caps_get_structure (copy, 0),
pos);
g_free (pos);
return copy;
}
}
}
}
gst_caps_unref (copy);
return NULL;
}
static GstElementStateReturn
gst_audio_convert_change_state (GstElement * element)
{
GstElementStateReturn ret;
GstAudioConvert *this = GST_AUDIO_CONVERT (element);
gint transition;
transition = GST_STATE_TRANSITION (element);
switch (transition) {
default:
break;
}
ret = parent_class->change_state (element);
switch (transition) {
case GST_STATE_PAUSED_TO_READY:
GST_STREAM_LOCK (this->sink);
this->convert_internal = NULL;
gst_audio_convert_unset_matrix (this);
gst_caps_replace (&GST_RPAD_CAPS (this->sink), NULL);
gst_caps_replace (&GST_RPAD_CAPS (this->src), NULL);
GST_STREAM_UNLOCK (this->sink);
break;
default:
break;
}
return ret;
}
/* 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;
g_assert (GST_IS_BUFFER (buf));
GST_LOG
("new buffer of size %u requested. Current is: data: %p - size: %u",
size, buf->data, buf->size);
if (buf->size >= 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",
buf->data, buf->size);
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", ret->data, ret->size);
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);
gst_buffer_set_caps (ret, GST_RPAD_CAPS (this->src));
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, guint8, 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 24:
{
/* Read 24-bits LE/BE into signed 64 host-endian */
if (this->sinkcaps.endianness == G_LITTLE_ENDIAN) {
cur = src[0] | (src[1] << 8) | (src[2] << 16);
} else {
cur = src[2] | (src[1] << 8) | (src[0] << 16);
}
/* Sign extend */
if ((this->sinkcaps.sign)
&& (cur & (1 << (this->sinkcaps.depth - 1))))
cur |= ((gint64) (-1)) ^ ((1 << this->sinkcaps.depth) - 1);
src -= 3;
}
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;
float temp;
/* 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);
gst_buffer_set_caps (ret, GST_RPAD_CAPS (this->src));
in = (gfloat *) GST_BUFFER_DATA (buf);
out = (gint32 *) GST_BUFFER_DATA (ret);
for (i = buf->size / sizeof (float); i > 0; i--) {
temp = *in * 2147483647.0f + .5;
*out = (gint32) CLAMP ((gint64) temp, -2147483648ll, 2147483647ll);
out++;
in++;
}
}
gst_buffer_unref (buf);
return ret;
}
#define POPULATE(out, format, be_func, le_func) G_STMT_START{ \
format val; \
format* p = (format *) out; \
int_value >>= (32 - this->srccaps.depth); \
if (this->srccaps.sign) { \
val = (format) int_value; \
} else { \
val = (format) int_value + (1 << (this->srccaps.depth - 1)); \
} \
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 ++; \
out = (guint8 *) p; \
}G_STMT_END
static GstBuffer *
gst_audio_convert_buffer_from_default_format (GstAudioConvert * this,
GstBuffer * buf)
{
GstBuffer *ret;
guint count, i;
gint32 *src;
if (this->srccaps.is_int && this->srccaps.width == 32
&& this->srccaps.depth == 32 && this->srccaps.endianness == G_BYTE_ORDER
&& this->srccaps.sign == TRUE)
return buf;
if (this->srccaps.is_int) {
guint8 *dest;
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);
gst_buffer_set_caps (ret, GST_RPAD_CAPS (this->src));
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 (dest, gint8, GINT8_IDENTITY, GINT8_IDENTITY);
} else {
POPULATE (dest, guint8, GUINT8_IDENTITY, GUINT8_IDENTITY);
}
break;
case 16:
if (this->srccaps.sign) {
POPULATE (dest, gint16, GINT16_TO_BE, GINT16_TO_LE);
} else {
POPULATE (dest, guint16, GUINT16_TO_BE, GUINT16_TO_LE);
}
break;
case 24:
{
guint8 tmp[4];
guint8 *tmpp = tmp;
/* Write out big endian array */
if (this->srccaps.sign) {
POPULATE (tmpp, gint32, GINT32_TO_BE, GINT32_TO_BE);
} else {
POPULATE (tmpp, guint32, GUINT32_TO_BE, GUINT32_TO_BE);
}
if (this->srccaps.endianness == G_LITTLE_ENDIAN) {
dest[2] = tmp[1];
dest[1] = tmp[2];
dest[0] = tmp[3];
} else {
memcpy (dest, tmp + 1, 3);
}
dest += 3;
}
break;
case 32:
if (this->srccaps.sign) {
POPULATE (dest, gint32, GINT32_TO_BE, GINT32_TO_LE);
} else {
POPULATE (dest, guint32, GUINT32_TO_BE, GUINT32_TO_LE);
}
break;
default:
g_assert_not_reached ();
}
}
} else {
gfloat *dest;
/* 1 / (2^31-1) * i */
#define INT2FLOAT(i) (4.6566128752457969e-10 * ((gfloat)i))
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);
gst_buffer_set_caps (ret, GST_RPAD_CAPS (this->src));
dest = (gfloat *) ret->data;
src = (gint32 *) buf->data;
for (i = 0; i < count; i++) {
*dest = (4.6566128752457969e-10 * ((gfloat) * src));
dest++;
src++;
}
}
gst_buffer_unref (buf);
return ret;
}
static GstBuffer *
gst_audio_convert_channels (GstAudioConvert * this, GstBuffer * buf)
{
GstBuffer *ret;
gint count;
g_assert (this->matrix != NULL);
/* check for passthrough */
if (gst_audio_convert_passthrough (this))
return buf;
/* convert */
count = GST_BUFFER_SIZE (buf) / 4 / this->sinkcaps.channels;
ret = gst_audio_convert_get_buffer (buf, count * 4 * this->srccaps.channels);
gst_buffer_set_caps (ret, GST_RPAD_CAPS (this->src));
gst_audio_convert_mix (this, (gint32 *) GST_BUFFER_DATA (buf),
(gint32 *) GST_BUFFER_DATA (ret), count);
gst_buffer_unref (buf);
return ret;
}
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