gstreamer/gst/audioconvert/gstaudioconvert.c
2012-02-27 12:52:07 +01:00

777 lines
24 KiB
C

/* GStreamer
* Copyright (C) 2003 Benjamin Otte <in7y118@public.uni-hamburg.de>
* Copyright (C) 2005 Thomas Vander Stichele <thomas at apestaart dot org>
* Copyright (C) 2011 Wim Taymans <wim.taymans at gmail 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,format=S8,channels=2 ! 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 GstCaps *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);
/* 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 (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL) \
", layout = (string) interleaved")
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 ******************************************************/
/* BaseTransform vmethods */
static gboolean
gst_audio_convert_get_unit_size (GstBaseTransform * base, GstCaps * caps,
gsize * size)
{
GstAudioInfo info;
g_assert (size);
if (!gst_audio_info_from_caps (&info, caps))
goto parse_error;
*size = info.bpf;
GST_INFO_OBJECT (base, "unit_size = %" G_GSIZE_FORMAT, *size);
return TRUE;
parse_error:
{
GST_INFO_OBJECT (base, "failed to parse caps to get unit_size");
return FALSE;
}
}
/* copies the given caps */
static GstCaps *
gst_audio_convert_caps_remove_format_info (GstCaps * caps)
{
GstStructure *st;
gint i, n;
GstCaps *res;
guint64 channel_mask;
res = gst_caps_new_empty ();
n = gst_caps_get_size (caps);
for (i = 0; i < n; i++) {
st = gst_caps_get_structure (caps, i);
/* If this is already expressed by the existing caps
* skip this structure */
if (i > 0 && gst_caps_is_subset_structure (res, st))
continue;
st = gst_structure_copy (st);
gst_structure_remove_field (st, "format");
/* Only remove the channels and channel-mask for non-NONE layouts */
if (gst_structure_get (st, "channel-mask", GST_TYPE_BITMASK, &channel_mask,
NULL)) {
if (channel_mask != 0)
gst_structure_remove_fields (st, "channel-mask", "channels", NULL);
} else {
gst_structure_remove_fields (st, "channel-mask", "channels", NULL);
}
gst_caps_append_structure (res, st);
}
return res;
}
/* The caps can be transformed into any other caps with format info removed.
* However, we should prefer passthrough, so if passthrough is possible,
* put it first in the list. */
static GstCaps *
gst_audio_convert_transform_caps (GstBaseTransform * btrans,
GstPadDirection direction, GstCaps * caps, GstCaps * filter)
{
GstCaps *tmp, *tmp2;
GstCaps *result;
/* Get all possible caps that we can transform to */
tmp = gst_audio_convert_caps_remove_format_info (caps);
if (filter) {
tmp2 = gst_caps_intersect_full (filter, tmp, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (tmp);
tmp = tmp2;
}
result = tmp;
GST_DEBUG_OBJECT (btrans, "transformed %" GST_PTR_FORMAT " into %"
GST_PTR_FORMAT, caps, result);
return result;
}
static const GstAudioChannelPosition default_positions[8][8] = {
/* 1 channel */
{
GST_AUDIO_CHANNEL_POSITION_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_LFE1,
},
/* 4 channels (4.0) */
{
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 (5.1) */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE1,
},
/* 7 channels (6.1) */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE1,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
},
/* 8 channels (7.1) */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE1,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
}
};
static gint
n_bits_set (guint64 x)
{
gint i;
gint c = 0;
guint64 y = 1;
for (i = 0; i < 64; i++) {
if (x & y)
c++;
y <<= 1;
}
return c;
}
static guint64
find_suitable_mask (guint64 mask, gint n_chans)
{
guint64 intersection;
gint i;
i = 0;
g_assert (n_bits_set (mask) >= n_chans);
intersection = mask;
do {
intersection = intersection & ((~G_GUINT64_CONSTANT (0)) >> i);
i++;
} while (n_bits_set (intersection) > n_chans && i < 64);
if (i < 64)
return intersection;
return 0;
}
static void
gst_audio_convert_fixate_channels (GstBaseTransform * base, GstStructure * ins,
GstStructure * outs)
{
gint in_chans, out_chans;
guint64 in_mask = 0, out_mask = 0;
gboolean has_in_mask = FALSE, has_out_mask = FALSE;
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-mask");
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-mask");
return;
}
/* get the channel layout of the output if any */
has_out_mask = gst_structure_has_field (outs, "channel-mask");
if (has_out_mask) {
gst_structure_get (outs, "channel-mask", GST_TYPE_BITMASK, &out_mask, NULL);
} else {
/* channels == 1 => MONO */
if (out_chans == 2) {
out_mask =
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT) |
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT);
has_out_mask = TRUE;
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, out_mask,
NULL);
}
}
/* get the channel layout of the input if any */
has_in_mask = gst_structure_has_field (ins, "channel-mask");
if (has_in_mask) {
gst_structure_get (ins, "channel-mask", GST_TYPE_BITMASK, &in_mask, NULL);
} else {
/* channels == 1 => MONO */
if (in_chans == 2) {
in_mask =
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT) |
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT);
has_in_mask = TRUE;
} else if (in_chans > 2)
g_warning ("%s: Upstream caps contain no channel mask",
GST_ELEMENT_NAME (base));
}
if (!has_out_mask && out_chans == 1 && (in_chans != out_chans
|| !has_in_mask))
return; /* nothing to do, default layout will be assumed */
if (in_chans == out_chans && (has_in_mask || in_chans == 1)) {
/* same number of channels and no output layout: just use input layout */
if (!has_out_mask) {
/* in_chans == 1 handled above already */
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, in_mask, NULL);
return;
}
/* If both masks are the same we're done, this includes the NONE layout case */
if (in_mask == out_mask)
return;
/* if output layout is fixed already and looks sane, we're done */
if (n_bits_set (out_mask) == out_chans)
return;
if (n_bits_set (out_mask) < in_chans) {
/* Not much we can do here, this shouldn't just happen */
g_warning ("%s: Invalid downstream channel-mask with too few bits set",
GST_ELEMENT_NAME (base));
} else {
guint64 intersection;
/* if the output layout is not fixed, check if the output layout contains
* the input layout */
intersection = in_mask & out_mask;
if (n_bits_set (intersection) >= in_chans) {
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, in_mask,
NULL);
return;
}
/* output layout is not fixed and does not contain the input layout, so
* just pick the first possibility */
intersection = find_suitable_mask (out_mask, out_chans);
if (intersection) {
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, intersection,
NULL);
return;
}
}
/* ... else fall back to default layout (NB: out_layout is NULL here) */
GST_WARNING_OBJECT (base, "unexpected output channel layout");
} else {
guint64 intersection;
/* 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 (n_bits_set (out_mask) >= out_chans) {
intersection = find_suitable_mask (out_mask, out_chans);
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, intersection,
NULL);
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])) {
gint i;
GST_DEBUG_OBJECT (base, "using default channel layout as fallback");
out_mask = 0;
for (i = 0; i < out_chans; i++)
out_mask |= G_GUINT64_CONSTANT (1) << default_positions[out_chans - 1][i];
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, out_mask, NULL);
} else {
GST_ERROR_OBJECT (base, "Have no default layout for %d channels",
out_chans);
}
}
/* 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 GstCaps *
gst_audio_convert_fixate_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * othercaps)
{
GstStructure *ins, *outs;
GstCaps *result;
GST_DEBUG_OBJECT (base, "trying to fixate othercaps %" GST_PTR_FORMAT
" based on caps %" GST_PTR_FORMAT, othercaps, caps);
result = gst_caps_intersect (othercaps, caps);
if (gst_caps_is_empty (result)) {
result = othercaps;
} else {
gst_caps_unref (othercaps);
}
/* fixate remaining fields */
result = gst_caps_make_writable (result);
ins = gst_caps_get_structure (caps, 0);
outs = gst_caps_get_structure (result, 0);
gst_audio_convert_fixate_channels (base, ins, outs);
gst_caps_fixate (result);
GST_DEBUG_OBJECT (base, "fixated othercaps to %" GST_PTR_FORMAT, result);
return result;
}
static gboolean
gst_audio_convert_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps)
{
GstAudioConvert *this = GST_AUDIO_CONVERT (base);
GstAudioInfo in_info;
GstAudioInfo out_info;
GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
if (!gst_audio_info_from_caps (&in_info, incaps))
goto invalid_in;
if (!gst_audio_info_from_caps (&out_info, outcaps))
goto invalid_out;
if (!audio_convert_prepare_context (&this->ctx, &in_info, &out_info,
this->dither, this->ns))
goto no_converter;
return TRUE;
/* ERRORS */
invalid_in:
{
GST_ERROR_OBJECT (base, "invalid input caps");
return FALSE;
}
invalid_out:
{
GST_ERROR_OBJECT (base, "invalid output caps");
return FALSE;
}
no_converter:
{
GST_ERROR_OBJECT (base, "could not find converter");
return FALSE;
}
}
static GstFlowReturn
gst_audio_convert_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
/* nothing to do here */
return GST_FLOW_OK;
}
static GstFlowReturn
gst_audio_convert_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstFlowReturn ret;
GstAudioConvert *this = GST_AUDIO_CONVERT (base);
GstMapInfo srcmap, dstmap;
gint insize, outsize;
gint samples;
/* get amount of samples to convert. */
samples = gst_buffer_get_size (inbuf) / this->ctx.in.bpf;
/* 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 */
gst_buffer_map (inbuf, &srcmap, GST_MAP_READ);
gst_buffer_map (outbuf, &dstmap, GST_MAP_WRITE);
/* check in and outsize */
if (srcmap.size < insize)
goto wrong_size;
if (dstmap.size < 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, srcmap.data, dstmap.data,
samples, gst_buffer_is_writable (inbuf)))
goto convert_error;
} else {
/* Create silence buffer */
gst_audio_format_fill_silence (this->ctx.out.finfo, dstmap.data, outsize);
}
ret = GST_FLOW_OK;
done:
gst_buffer_unmap (outbuf, &dstmap);
gst_buffer_unmap (inbuf, &srcmap);
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: %" G_GSIZE_FORMAT " < %d"
" or out: %" G_GSIZE_FORMAT " < %d",
srcmap.size, insize, dstmap.size, 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;
}
}