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gstreamer/gst/audioconvert/gstaudioconvert.c
Sebastian Dröge 4fcd621101 audioconvert: Use new gst_caps_is_subset_structure() API 
This prevents one copy of every structure and creating a new caps
instance.
2011-05-27 14:10:50 +02:00

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/* GStreamer
* Copyright (C) 2003 Benjamin Otte <in7y118@public.uni-hamburg.de>
* Copyright (C) 2005 Thomas Vander Stichele <thomas at apestaart dot org>
* Copyright (C) 2005 Wim Taymans <wim at fluendo dot com>
*
* 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.
*/
/**
* SECTION:element-audioconvert
*
* Audioconvert converts raw audio buffers between various possible formats.
* It supports integer to float conversion, width/depth conversion,
* signedness and endianness conversion and channel transformations.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch -v -m audiotestsrc ! audioconvert ! audio/x-raw-int,channels=2,width=8,depth=8 ! level ! fakesink silent=TRUE
* ]| This pipeline converts audio to 8-bit. The level element shows that
* the output levels still match the one for a sine wave.
* |[
* gst-launch -v -m audiotestsrc ! audioconvert ! vorbisenc ! fakesink silent=TRUE
* ]| The vorbis encoder takes float audio data instead of the integer data
* generated by audiotestsrc.
* </refsect2>
*
* Last reviewed on 2006-03-02 (0.10.4)
*/
/*
* 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 <string.h>
#include "gstaudioconvert.h"
#include "gstchannelmix.h"
#include "gstaudioquantize.h"
#include "plugin.h"
GST_DEBUG_CATEGORY (audio_convert_debug);
GST_DEBUG_CATEGORY_STATIC (GST_CAT_PERFORMANCE);
/*** DEFINITIONS **************************************************************/
/* type functions */
static void gst_audio_convert_dispose (GObject * obj);
/* gstreamer functions */
static gboolean gst_audio_convert_get_unit_size (GstBaseTransform * base,
GstCaps * caps, gsize * size);
static GstCaps *gst_audio_convert_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * filter);
static void gst_audio_convert_fixate_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * othercaps);
static gboolean gst_audio_convert_set_caps (GstBaseTransform * base,
GstCaps * incaps, GstCaps * outcaps);
static GstFlowReturn gst_audio_convert_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
static GstFlowReturn gst_audio_convert_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
static void gst_audio_convert_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_audio_convert_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean structure_has_fixed_channel_positions (GstStructure * s,
gboolean * unpositioned_layout);
/* AudioConvert signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_DITHERING,
ARG_NOISE_SHAPING,
};
#define DEBUG_INIT \
GST_DEBUG_CATEGORY_INIT (audio_convert_debug, "audioconvert", 0, "audio conversion element"); \
GST_DEBUG_CATEGORY_GET (GST_CAT_PERFORMANCE, "GST_PERFORMANCE");
#define gst_audio_convert_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstAudioConvert, gst_audio_convert,
GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
/*** GSTREAMER PROTOTYPES *****************************************************/
#define STATIC_CAPS \
GST_STATIC_CAPS ( \
"audio/x-raw-float, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 64;" \
"audio/x-raw-float, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 32;" \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 32, " \
"depth = (int) [ 1, 32 ], " \
"signed = (boolean) { true, false }; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"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, MAX ], " \
"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, MAX ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 8, " \
"depth = (int) [ 1, 8 ], " \
"signed = (boolean) { true, false } " \
)
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);
#define GST_TYPE_AUDIO_CONVERT_DITHERING (gst_audio_convert_dithering_get_type ())
static GType
gst_audio_convert_dithering_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{DITHER_NONE, "No dithering",
"none"},
{DITHER_RPDF, "Rectangular dithering", "rpdf"},
{DITHER_TPDF, "Triangular dithering (default)", "tpdf"},
{DITHER_TPDF_HF, "High frequency triangular dithering", "tpdf-hf"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioConvertDithering", values);
}
return gtype;
}
#define GST_TYPE_AUDIO_CONVERT_NOISE_SHAPING (gst_audio_convert_ns_get_type ())
static GType
gst_audio_convert_ns_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{NOISE_SHAPING_NONE, "No noise shaping (default)",
"none"},
{NOISE_SHAPING_ERROR_FEEDBACK, "Error feedback", "error-feedback"},
{NOISE_SHAPING_SIMPLE, "Simple 2-pole noise shaping", "simple"},
{NOISE_SHAPING_MEDIUM, "Medium 5-pole noise shaping", "medium"},
{NOISE_SHAPING_HIGH, "High 8-pole noise shaping", "high"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioConvertNoiseShaping", values);
}
return gtype;
}
/*** TYPE FUNCTIONS ***********************************************************/
static void
gst_audio_convert_class_init (GstAudioConvertClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseTransformClass *basetransform_class = GST_BASE_TRANSFORM_CLASS (klass);
gobject_class->dispose = gst_audio_convert_dispose;
gobject_class->set_property = gst_audio_convert_set_property;
gobject_class->get_property = gst_audio_convert_get_property;
g_object_class_install_property (gobject_class, ARG_DITHERING,
g_param_spec_enum ("dithering", "Dithering",
"Selects between different dithering methods.",
GST_TYPE_AUDIO_CONVERT_DITHERING, DITHER_TPDF,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, ARG_NOISE_SHAPING,
g_param_spec_enum ("noise-shaping", "Noise shaping",
"Selects between different noise shaping methods.",
GST_TYPE_AUDIO_CONVERT_NOISE_SHAPING, NOISE_SHAPING_NONE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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_simple (element_class,
"Audio converter", "Filter/Converter/Audio",
"Convert audio to different formats", "Benjamin Otte <otte@gnome.org>");
basetransform_class->get_unit_size =
GST_DEBUG_FUNCPTR (gst_audio_convert_get_unit_size);
basetransform_class->transform_caps =
GST_DEBUG_FUNCPTR (gst_audio_convert_transform_caps);
basetransform_class->fixate_caps =
GST_DEBUG_FUNCPTR (gst_audio_convert_fixate_caps);
basetransform_class->set_caps =
GST_DEBUG_FUNCPTR (gst_audio_convert_set_caps);
basetransform_class->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_convert_transform_ip);
basetransform_class->transform =
GST_DEBUG_FUNCPTR (gst_audio_convert_transform);
basetransform_class->passthrough_on_same_caps = TRUE;
}
static void
gst_audio_convert_init (GstAudioConvert * this)
{
this->dither = DITHER_TPDF;
this->ns = NOISE_SHAPING_NONE;
memset (&this->ctx, 0, sizeof (AudioConvertCtx));
gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (this), TRUE);
}
static void
gst_audio_convert_dispose (GObject * obj)
{
GstAudioConvert *this = GST_AUDIO_CONVERT (obj);
audio_convert_clean_context (&this->ctx);
G_OBJECT_CLASS (parent_class)->dispose (obj);
}
/*** GSTREAMER FUNCTIONS ******************************************************/
/* convert the given GstCaps to our format */
static gboolean
gst_audio_convert_parse_caps (const GstCaps * caps, AudioConvertFmt * fmt)
{
GstStructure *structure = gst_caps_get_structure (caps, 0);
GST_DEBUG ("parse caps %p and %" GST_PTR_FORMAT, caps, caps);
g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
g_return_val_if_fail (fmt != NULL, FALSE);
/* cleanup old */
audio_convert_clean_fmt (fmt);
fmt->endianness = G_BYTE_ORDER;
fmt->is_int =
(strcmp (gst_structure_get_name (structure), "audio/x-raw-int") == 0);
/* parse common fields */
if (!gst_structure_get_int (structure, "channels", &fmt->channels))
goto no_values;
if (!(fmt->pos = gst_audio_get_channel_positions (structure)))
goto no_values;
fmt->unpositioned_layout = FALSE;
structure_has_fixed_channel_positions (structure, &fmt->unpositioned_layout);
if (!gst_structure_get_int (structure, "width", &fmt->width))
goto no_values;
if (!gst_structure_get_int (structure, "rate", &fmt->rate))
goto no_values;
/* width != 8 needs an endianness field */
if (fmt->width != 8) {
if (!gst_structure_get_int (structure, "endianness", &fmt->endianness))
goto no_values;
}
if (fmt->is_int) {
/* int specific fields */
if (!gst_structure_get_boolean (structure, "signed", &fmt->sign))
goto no_values;
if (!gst_structure_get_int (structure, "depth", &fmt->depth))
goto no_values;
/* depth cannot be bigger than the width */
if (fmt->depth > fmt->width)
goto not_allowed;
}
fmt->unit_size = (fmt->width * fmt->channels) / 8;
return TRUE;
/* ERRORS */
no_values:
{
GST_DEBUG ("could not get some values from structure");
audio_convert_clean_fmt (fmt);
return FALSE;
}
not_allowed:
{
GST_DEBUG ("width > depth, not allowed - make us advertise correct fmt");
audio_convert_clean_fmt (fmt);
return FALSE;
}
}
/* BaseTransform vmethods */
static gboolean
gst_audio_convert_get_unit_size (GstBaseTransform * base, GstCaps * caps,
gsize * size)
{
AudioConvertFmt fmt = { 0 };
g_assert (size);
if (!gst_audio_convert_parse_caps (caps, &fmt))
goto parse_error;
GST_INFO_OBJECT (base, "unit_size = %u", fmt.unit_size);
*size = fmt.unit_size;
audio_convert_clean_fmt (&fmt);
return TRUE;
parse_error:
{
GST_INFO_OBJECT (base, "failed to parse caps to get unit_size");
return FALSE;
}
}
/* Set widths (a list); multiples of 8 between min and max */
static void
set_structure_widths (GstStructure * s, int min, int max)
{
GValue list = { 0 };
GValue val = { 0 };
int width;
if (min == max) {
gst_structure_set (s, "width", G_TYPE_INT, min, NULL);
return;
}
g_value_init (&list, GST_TYPE_LIST);
g_value_init (&val, G_TYPE_INT);
for (width = min; width <= max; width += 8) {
g_value_set_int (&val, width);
gst_value_list_append_value (&list, &val);
}
gst_structure_set_value (s, "width", &list);
g_value_unset (&val);
g_value_unset (&list);
}
/* Set widths of 32 bits and 64 bits (as list) */
static void
set_structure_widths_32_and_64 (GstStructure * s)
{
GValue list = { 0 };
GValue val = { 0 };
g_value_init (&list, GST_TYPE_LIST);
g_value_init (&val, G_TYPE_INT);
g_value_set_int (&val, 32);
gst_value_list_append_value (&list, &val);
g_value_set_int (&val, 64);
gst_value_list_append_value (&list, &val);
gst_structure_set_value (s, "width", &list);
g_value_unset (&val);
g_value_unset (&list);
}
/* Modify the structure so that things that must always have a single
* value (for float), or can always be losslessly converted (for int), have
* appropriate values.
*/
static GstStructure *
make_lossless_changes (GstStructure * s, gboolean isfloat)
{
GValue list = { 0 };
GValue val = { 0 };
int i;
const gint endian[] = { G_LITTLE_ENDIAN, G_BIG_ENDIAN };
const gboolean booleans[] = { TRUE, FALSE };
g_value_init (&list, GST_TYPE_LIST);
g_value_init (&val, G_TYPE_INT);
for (i = 0; i < 2; i++) {
g_value_set_int (&val, endian[i]);
gst_value_list_append_value (&list, &val);
}
gst_structure_set_value (s, "endianness", &list);
g_value_unset (&val);
g_value_unset (&list);
if (isfloat) {
/* float doesn't have a depth or signedness field and only supports
* widths of 32 and 64 bits */
gst_structure_remove_field (s, "depth");
gst_structure_remove_field (s, "signed");
set_structure_widths_32_and_64 (s);
} else {
/* int supports signed and unsigned. GValues are a pain */
g_value_init (&list, GST_TYPE_LIST);
g_value_init (&val, G_TYPE_BOOLEAN);
for (i = 0; i < 2; i++) {
g_value_set_boolean (&val, booleans[i]);
gst_value_list_append_value (&list, &val);
}
gst_structure_set_value (s, "signed", &list);
g_value_unset (&val);
g_value_unset (&list);
}
return s;
}
static void
strip_width_64 (GstStructure * s)
{
const GValue *v = gst_structure_get_value (s, "width");
GValue widths = { 0 };
if (GST_VALUE_HOLDS_LIST (v)) {
int i;
int len = gst_value_list_get_size (v);
g_value_init (&widths, GST_TYPE_LIST);
for (i = 0; i < len; i++) {
const GValue *width = gst_value_list_get_value (v, i);
if (g_value_get_int (width) != 64)
gst_value_list_append_value (&widths, width);
}
gst_structure_set_value (s, "width", &widths);
g_value_unset (&widths);
}
}
/* Little utility function to create a related structure for float/int */
static void
append_with_other_format (GstCaps * caps, const GstStructure * s,
gboolean isfloat)
{
GstStructure *s2;
if (isfloat) {
s2 = gst_structure_copy (s);
gst_structure_set_name (s2, "audio/x-raw-int");
make_lossless_changes (s2, FALSE);
/* If 64 bit float was allowed; remove width 64: we don't support it for
* integer*/
strip_width_64 (s2);
gst_caps_merge_structure (caps, s2);
} else {
s2 = gst_structure_copy (s);
gst_structure_set_name (s2, "audio/x-raw-float");
make_lossless_changes (s2, TRUE);
gst_caps_merge_structure (caps, s2);
}
}
static gboolean
structure_has_fixed_channel_positions (GstStructure * s,
gboolean * unpositioned_layout)
{
GstAudioChannelPosition *pos;
const GValue *val;
gint channels = 0;
if (!gst_structure_get_int (s, "channels", &channels))
return FALSE; /* probably a range */
val = gst_structure_get_value (s, "channel-positions");
if ((val == NULL || !gst_value_is_fixed (val)) && channels <= 8) {
GST_LOG ("no or unfixed channel-positions in %" GST_PTR_FORMAT, s);
return FALSE;
} else if (val == NULL || !gst_value_is_fixed (val)) {
GST_LOG ("implicit undefined channel-positions");
*unpositioned_layout = TRUE;
return TRUE;
}
pos = gst_audio_get_channel_positions (s);
if (pos && pos[0] == GST_AUDIO_CHANNEL_POSITION_NONE) {
GST_LOG ("fixed undefined channel-positions in %" GST_PTR_FORMAT, s);
*unpositioned_layout = TRUE;
} else {
GST_LOG ("fixed defined channel-positions in %" GST_PTR_FORMAT, s);
*unpositioned_layout = FALSE;
}
g_free (pos);
return TRUE;
}
/* Audioconvert can perform all conversions on audio except for resampling.
* However, there are some conversions we _prefer_ not to do. For example, it's
* better to convert format (float<->int, endianness, etc) than the number of
* channels, as the latter conversion is not lossless.
*
* So, we return, in order (assuming input caps have only one structure;
* which is enforced by basetransform):
* - input caps with a different format (lossless conversions).
* - input caps with a different format (slightly lossy conversions).
* - input caps with a different number of channels (very lossy!)
*/
static GstCaps *
gst_audio_convert_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * filter)
{
GstCaps *ret;
GstStructure *s, *structure;
gboolean isfloat, allow_mixing;
gint width, depth, channels = 0;
const gchar *fields_used[] = {
"width", "depth", "rate", "channels", "endianness", "signed"
};
const gchar *structure_name;
gint n, j;
int i;
n = gst_caps_get_size (caps);
ret = gst_caps_new_empty ();
for (j = 0; j < n; j++) {
structure = gst_caps_get_structure (caps, j);
if (j > 0) {
/* If the new structure is a subset of the already existing transformed
* caps we can safely skip it because we would transform it to the
* same caps again.
*/
if (gst_caps_is_subset_structure (ret, structure))
continue;
}
structure_name = gst_structure_get_name (structure);
isfloat = strcmp (structure_name, "audio/x-raw-float") == 0;
/* We operate on a version of the original structure with any additional
* fields absent */
s = gst_structure_empty_new (structure_name);
for (i = 0; i < sizeof (fields_used) / sizeof (*fields_used); i++) {
if (gst_structure_has_field (structure, fields_used[i]))
gst_structure_set_value (s, fields_used[i],
gst_structure_get_value (structure, fields_used[i]));
}
if (!isfloat) {
/* Commonly, depth is left out: set it equal to width if we have a fixed
* width, if so */
if (!gst_structure_has_field (s, "depth") &&
gst_structure_get_int (s, "width", &width))
gst_structure_set (s, "depth", G_TYPE_INT, width, NULL);
}
/* All lossless conversions */
s = make_lossless_changes (s, isfloat);
gst_caps_merge_structure (ret, gst_structure_copy (s));
/* Same, plus a float<->int conversion */
append_with_other_format (ret, s, isfloat);
GST_DEBUG_OBJECT (base, " step1: (%d) %" GST_PTR_FORMAT,
gst_caps_get_size (ret), ret);
/* We don't mind increasing width/depth/channels, but reducing them is
* Very Bad. Only available if width, depth, channels are already fixed. */
if (!isfloat) {
if (gst_structure_get_int (structure, "width", &width))
set_structure_widths (s, width, 32);
if (gst_structure_get_int (structure, "depth", &depth)) {
if (depth == 32)
gst_structure_set (s, "depth", G_TYPE_INT, 32, NULL);
else
gst_structure_set (s, "depth", GST_TYPE_INT_RANGE, depth, 32, NULL);
}
}
allow_mixing = TRUE;
if (gst_structure_get_int (structure, "channels", &channels)) {
gboolean unpositioned;
/* we don't support mixing for channels without channel positions */
if (structure_has_fixed_channel_positions (structure, &unpositioned))
allow_mixing = (unpositioned == FALSE);
}
if (!allow_mixing) {
gst_structure_set (s, "channels", G_TYPE_INT, channels, NULL);
if (gst_structure_has_field (structure, "channel-positions"))
gst_structure_set_value (s, "channel-positions",
gst_structure_get_value (structure, "channel-positions"));
} else {
if (channels == 0)
gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, 1, 11, NULL);
else if (channels == 11)
gst_structure_set (s, "channels", G_TYPE_INT, 11, NULL);
else
gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, channels, 11,
NULL);
gst_structure_remove_field (s, "channel-positions");
}
gst_caps_merge_structure (ret, gst_structure_copy (s));
/* Same, plus a float<->int conversion */
append_with_other_format (ret, s, isfloat);
/* We'll reduce depth if we must. We reduce as low as 16 bits (for integer);
* reducing to less than this is even worse than dropping channels. We only
* do this if we haven't already done the equivalent above. */
if (!gst_structure_get_int (structure, "width", &width) || width > 16) {
if (isfloat) {
GstStructure *s2 = gst_structure_copy (s);
set_structure_widths_32_and_64 (s2);
append_with_other_format (ret, s2, TRUE);
gst_structure_free (s2);
} else {
GstStructure *s2 = gst_structure_copy (s);
set_structure_widths (s2, 16, 32);
gst_structure_set (s2, "depth", GST_TYPE_INT_RANGE, 16, 32, NULL);
gst_caps_merge_structure (ret, s2);
}
}
/* Channel conversions to fewer channels is only done if needed - generally
* it's very bad to drop channels entirely.
*/
if (allow_mixing) {
gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, 1, 11, NULL);
gst_structure_remove_field (s, "channel-positions");
} else {
/* allow_mixing can only be FALSE if we got a fixed number of channels */
gst_structure_set (s, "channels", G_TYPE_INT, channels, NULL);
if (gst_structure_has_field (structure, "channel-positions"))
gst_structure_set_value (s, "channel-positions",
gst_structure_get_value (structure, "channel-positions"));
}
gst_caps_merge_structure (ret, gst_structure_copy (s));
/* Same, plus a float<->int conversion */
append_with_other_format (ret, s, isfloat);
/* And, finally, for integer only, we allow conversion to any width/depth we
* support: this should be equivalent to our (non-float) template caps. (the
* floating point case should be being handled just above) */
set_structure_widths (s, 8, 32);
gst_structure_set (s, "depth", GST_TYPE_INT_RANGE, 1, 32, NULL);
if (isfloat) {
append_with_other_format (ret, s, TRUE);
gst_structure_free (s);
} else
gst_caps_merge_structure (ret, s);
}
GST_DEBUG_OBJECT (base, "Caps transformed to %" GST_PTR_FORMAT, ret);
if (filter) {
GstCaps *intersection;
GST_DEBUG_OBJECT (base, "Using filter caps %" GST_PTR_FORMAT, filter);
intersection =
gst_caps_intersect_full (filter, ret, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (ret);
ret = intersection;
GST_DEBUG_OBJECT (base, "Intersection %" GST_PTR_FORMAT, ret);
}
return ret;
}
static const GstAudioChannelPosition default_positions[8][8] = {
/* 1 channel */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_MONO,
},
/* 2 channels */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
},
/* 3 channels (2.1) */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE, /* or FRONT_CENTER for 3.0? */
},
/* 4 channels (4.0 or 3.1?) */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
},
/* 5 channels */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
},
/* 6 channels */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
},
/* 7 channels */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
},
/* 8 channels */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
}
};
static const GValue *
find_suitable_channel_layout (const GValue * val, guint chans)
{
/* if output layout is fixed already and looks sane, we're done */
if (GST_VALUE_HOLDS_ARRAY (val) && gst_value_array_get_size (val) == chans)
return val;
/* if it's a list, go through it recursively and return the first
* sane-enough looking value we find */
if (GST_VALUE_HOLDS_LIST (val)) {
gint i;
for (i = 0; i < gst_value_list_get_size (val); ++i) {
const GValue *v, *ret;
v = gst_value_list_get_value (val, i);
if ((ret = find_suitable_channel_layout (v, chans)))
return ret;
}
}
return NULL;
}
static void
gst_audio_convert_fixate_channels (GstBaseTransform * base, GstStructure * ins,
GstStructure * outs)
{
const GValue *in_layout, *out_layout;
gint in_chans, out_chans;
if (!gst_structure_get_int (ins, "channels", &in_chans))
return; /* this shouldn't really happen, should it? */
if (!gst_structure_has_field (outs, "channels")) {
/* we could try to get the implied number of channels from the layout,
* but that seems overdoing it for a somewhat exotic corner case */
gst_structure_remove_field (outs, "channel-positions");
return;
}
/* ok, let's fixate the channels if they are not fixated yet */
gst_structure_fixate_field_nearest_int (outs, "channels", in_chans);
if (!gst_structure_get_int (outs, "channels", &out_chans)) {
/* shouldn't really happen ... */
gst_structure_remove_field (outs, "channel-positions");
return;
}
/* check if the output has a channel layout (or a list of layouts) */
out_layout = gst_structure_get_value (outs, "channel-positions");
/* get the channel layout of the input if any */
in_layout = gst_structure_get_value (ins, "channel-positions");
if (out_layout == NULL) {
if (out_chans <= 2 && (in_chans != out_chans || in_layout == NULL))
return; /* nothing to do, default layout will be assumed */
GST_WARNING_OBJECT (base, "downstream caps contain no channel layout");
}
if (in_chans == out_chans && in_layout != NULL) {
GValue res = { 0, };
/* same number of channels and no output layout: just use input layout */
if (out_layout == NULL) {
gst_structure_set_value (outs, "channel-positions", in_layout);
return;
}
/* if output layout is fixed already and looks sane, we're done */
if (GST_VALUE_HOLDS_ARRAY (out_layout) &&
gst_value_array_get_size (out_layout) == out_chans) {
return;
}
/* if the output layout is not fixed, check if the output layout contains
* the input layout */
if (gst_value_intersect (&res, in_layout, out_layout)) {
gst_structure_set_value (outs, "channel-positions", in_layout);
g_value_unset (&res);
return;
}
/* output layout is not fixed and does not contain the input layout, so
* just pick the first layout in the list (it should be a list ...) */
if ((out_layout = find_suitable_channel_layout (out_layout, out_chans))) {
gst_structure_set_value (outs, "channel-positions", out_layout);
return;
}
/* ... else fall back to default layout (NB: out_layout is NULL here) */
GST_WARNING_OBJECT (base, "unexpected output channel layout");
}
/* number of input channels != number of output channels:
* if this value contains a list of channel layouts (or even worse: a list
* with another list), just pick the first value and repeat until we find a
* channel position array or something else that's not a list; we assume
* the input if half-way sane and don't try to fall back on other list items
* if the first one is something unexpected or non-channel-pos-array-y */
if (out_layout != NULL && GST_VALUE_HOLDS_LIST (out_layout))
out_layout = find_suitable_channel_layout (out_layout, out_chans);
if (out_layout != NULL) {
if (GST_VALUE_HOLDS_ARRAY (out_layout) &&
gst_value_array_get_size (out_layout) == out_chans) {
/* looks sane enough, let's use it */
gst_structure_set_value (outs, "channel-positions", out_layout);
return;
}
/* what now?! Just ignore what we're given and use default positions */
GST_WARNING_OBJECT (base, "invalid or unexpected channel-positions");
}
/* missing or invalid output layout and we can't use the input layout for
* one reason or another, so just pick a default layout (we could be smarter
* and try to add/remove channels from the input layout, or pick a default
* layout based on LFE-presence in input layout, but let's save that for
* another day) */
if (out_chans > 0 && out_chans <= G_N_ELEMENTS (default_positions[0])) {
GST_DEBUG_OBJECT (base, "using default channel layout as fallback");
gst_audio_set_channel_positions (outs, default_positions[out_chans - 1]);
}
}
/* try to keep as many of the structure members the same by fixating the
* possible ranges; this way we convert the least amount of things as possible
*/
static void
gst_audio_convert_fixate_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * othercaps)
{
GstStructure *ins, *outs;
gint rate, endianness, depth, width;
gboolean signedness;
g_return_if_fail (gst_caps_is_fixed (caps));
GST_DEBUG_OBJECT (base, "trying to fixate othercaps %" GST_PTR_FORMAT
" based on caps %" GST_PTR_FORMAT, othercaps, caps);
ins = gst_caps_get_structure (caps, 0);
outs = gst_caps_get_structure (othercaps, 0);
gst_audio_convert_fixate_channels (base, ins, outs);
if (gst_structure_get_int (ins, "rate", &rate)) {
if (gst_structure_has_field (outs, "rate")) {
gst_structure_fixate_field_nearest_int (outs, "rate", rate);
}
}
if (gst_structure_get_int (ins, "endianness", &endianness)) {
if (gst_structure_has_field (outs, "endianness")) {
gst_structure_fixate_field_nearest_int (outs, "endianness", endianness);
}
}
if (gst_structure_get_int (ins, "width", &width)) {
if (gst_structure_has_field (outs, "width")) {
gst_structure_fixate_field_nearest_int (outs, "width", width);
}
} else {
/* this is not allowed */
}
if (gst_structure_get_int (ins, "depth", &depth)) {
if (gst_structure_has_field (outs, "depth")) {
gst_structure_fixate_field_nearest_int (outs, "depth", depth);
}
} else {
/* set depth as width */
if (gst_structure_has_field (outs, "depth")) {
gst_structure_fixate_field_nearest_int (outs, "depth", width);
}
}
if (gst_structure_get_boolean (ins, "signed", &signedness)) {
if (gst_structure_has_field (outs, "signed")) {
gst_structure_fixate_field_boolean (outs, "signed", signedness);
}
}
GST_DEBUG_OBJECT (base, "fixated othercaps to %" GST_PTR_FORMAT, othercaps);
}
static gboolean
gst_audio_convert_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps)
{
AudioConvertFmt in_ac_caps = { 0 };
AudioConvertFmt out_ac_caps = { 0 };
GstAudioConvert *this = GST_AUDIO_CONVERT (base);
GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
if (!gst_audio_convert_parse_caps (incaps, &in_ac_caps))
return FALSE;
if (!gst_audio_convert_parse_caps (outcaps, &out_ac_caps))
return FALSE;
if (!audio_convert_prepare_context (&this->ctx, &in_ac_caps, &out_ac_caps,
this->dither, this->ns))
goto no_converter;
return TRUE;
no_converter:
{
return FALSE;
}
}
static GstFlowReturn
gst_audio_convert_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
/* nothing to do here */
return GST_FLOW_OK;
}
static void
gst_audio_convert_create_silence_buffer (GstAudioConvert * this, gpointer dst,
gint size)
{
if (this->ctx.out.is_int && !this->ctx.out.sign) {
gint i;
switch (this->ctx.out.width) {
case 8:{
guint8 zero = 0x80 >> (8 - this->ctx.out.depth);
memset (dst, zero, size);
break;
}
case 16:{
guint16 *data = (guint16 *) dst;
guint16 zero = 0x8000 >> (16 - this->ctx.out.depth);
if (this->ctx.out.endianness == G_LITTLE_ENDIAN)
zero = GUINT16_TO_LE (zero);
else
zero = GUINT16_TO_BE (zero);
size /= 2;
for (i = 0; i < size; i++)
data[i] = zero;
break;
}
case 24:{
guint32 zero = 0x800000 >> (24 - this->ctx.out.depth);
guint8 *data = (guint8 *) dst;
if (this->ctx.out.endianness == G_LITTLE_ENDIAN) {
for (i = 0; i < size; i += 3) {
data[i] = zero & 0xff;
data[i + 1] = (zero >> 8) & 0xff;
data[i + 2] = (zero >> 16) & 0xff;
}
} else {
for (i = 0; i < size; i += 3) {
data[i + 2] = zero & 0xff;
data[i + 1] = (zero >> 8) & 0xff;
data[i] = (zero >> 16) & 0xff;
}
}
break;
}
case 32:{
guint32 *data = (guint32 *) dst;
guint32 zero = (0x80000000 >> (32 - this->ctx.out.depth));
if (this->ctx.out.endianness == G_LITTLE_ENDIAN)
zero = GUINT32_TO_LE (zero);
else
zero = GUINT32_TO_BE (zero);
size /= 4;
for (i = 0; i < size; i++)
data[i] = zero;
break;
}
default:
memset (dst, 0, size);
g_return_if_reached ();
break;
}
} else {
memset (dst, 0, size);
}
}
static GstFlowReturn
gst_audio_convert_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstFlowReturn ret;
GstAudioConvert *this = GST_AUDIO_CONVERT (base);
gsize srcsize, dstsize;
gint insize, outsize;
gint samples;
gpointer src, dst;
/* get amount of samples to convert. */
samples = gst_buffer_get_size (inbuf) / this->ctx.in.unit_size;
/* get in/output sizes, to see if the buffers we got are of correct
* sizes */
if (!audio_convert_get_sizes (&this->ctx, samples, &insize, &outsize))
goto error;
if (insize == 0 || outsize == 0)
return GST_FLOW_OK;
/* get src and dst data */
src = gst_buffer_map (inbuf, &srcsize, NULL, GST_MAP_READ);
dst = gst_buffer_map (outbuf, &dstsize, NULL, GST_MAP_WRITE);
/* check in and outsize */
if (srcsize < insize)
goto wrong_size;
if (dstsize < outsize)
goto wrong_size;
/* and convert the samples */
if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
if (!audio_convert_convert (&this->ctx, src, dst,
samples, gst_buffer_is_writable (inbuf)))
goto convert_error;
} else {
/* Create silence buffer */
gst_audio_convert_create_silence_buffer (this, dst, outsize);
}
ret = GST_FLOW_OK;
done:
gst_buffer_unmap (outbuf, dst, outsize);
gst_buffer_unmap (inbuf, src, srcsize);
return ret;
/* ERRORS */
error:
{
GST_ELEMENT_ERROR (this, STREAM, FORMAT,
(NULL), ("cannot get input/output sizes for %d samples", samples));
return GST_FLOW_ERROR;
}
wrong_size:
{
GST_ELEMENT_ERROR (this, STREAM, FORMAT,
(NULL),
("input/output buffers are of wrong size in: %d < %d or out: %d < %d",
srcsize, insize, dstsize, outsize));
ret = GST_FLOW_ERROR;
goto done;
}
convert_error:
{
GST_ELEMENT_ERROR (this, STREAM, FORMAT,
(NULL), ("error while converting"));
ret = GST_FLOW_ERROR;
goto done;
}
}
static void
gst_audio_convert_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioConvert *this = GST_AUDIO_CONVERT (object);
switch (prop_id) {
case ARG_DITHERING:
this->dither = g_value_get_enum (value);
break;
case ARG_NOISE_SHAPING:
this->ns = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_convert_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioConvert *this = GST_AUDIO_CONVERT (object);
switch (prop_id) {
case ARG_DITHERING:
g_value_set_enum (value, this->dither);
break;
case ARG_NOISE_SHAPING:
g_value_set_enum (value, this->ns);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
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