/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-audioconvert
* @title: 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
* (ie. upmixing and downmixing), as well as dithering and noise-shaping.
*
* ## Example launch line
* |[
* gst-launch-1.0 -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-1.0 -v -m uridecodebin uri=file:///path/to/audio.flac ! audioconvert ! vorbisenc ! oggmux ! filesink location=audio.ogg
* ]|
* The vorbis encoder takes float audio data instead of the integer data
* output by most other audio elements. This pipeline decodes a FLAC audio file
* (or any other audio file for which decoders are installed) and re-encodes
* it into an Ogg/Vorbis audio file.
*
* A mix matrix can be passed to audioconvert, that will govern the
* remapping of input to output channels.
* ## Example matrix generation code
* To generate the matrix using code:
*
* |[
* GValue v = G_VALUE_INIT;
* GValue v2 = G_VALUE_INIT;
* GValue v3 = G_VALUE_INIT;
*
* g_value_init (&v2, GST_TYPE_ARRAY);
* g_value_init (&v3, G_TYPE_FLOAT);
* g_value_set_float (&v3, 1);
* gst_value_array_append_value (&v2, &v3);
* g_value_unset (&v3);
* [ Repeat for as many float as your input channels - unset and reinit v3 ]
* g_value_init (&v, GST_TYPE_ARRAY);
* gst_value_array_append_value (&v, &v2);
* g_value_unset (&v2);
* [ Repeat for as many v2's as your output channels - unset and reinit v2]
* g_object_set_property (G_OBJECT (audioconvert), "mix-matrix", &v);
* g_value_unset (&v);
* ]|
*
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc ! audio/x-raw, channels=4 ! audioconvert mix-matrix="<<(float)1.0, (float)0.0, (float)0.0, (float)0.0>, <(float)0.0, (float)1.0, (float)0.0, (float)0.0>>" ! audio/x-raw,channels=2 ! autoaudiosink
* ]|
*
* > If an empty mix matrix is specified, a (potentially truncated)
* > identity matrix will be generated.
*
* ## Example empty matrix generation code
* |[
* GValue v = G_VALUE_INIT;
*
* g_value_init (&v, GST_TYPE_ARRAY);
* g_object_set_property (G_OBJECT (audioconvert), "mix-matrix", &v);
* g_value_unset (&v);
* ]|
*
* ## Example empty matrix launch line
* |[
* gst-launch-1.0 -v audiotestsrc ! audio/x-raw,channels=8 ! audioconvert mix-matrix="<>" ! audio/x-raw,channels=16,channel-mask=\(bitmask\)0x0000000000000000 ! fakesink
* ]|
*/
/*
* 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 "plugin.h"
GST_DEBUG_CATEGORY (audio_convert_debug);
GST_DEBUG_CATEGORY_STATIC (GST_CAT_PERFORMANCE);
#define GST_CAT_DEFAULT (audio_convert_debug)
/*** 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 gboolean gst_audio_convert_transform_meta (GstBaseTransform * trans,
GstBuffer * outbuf, GstMeta * meta, GstBuffer * inbuf);
static GstFlowReturn gst_audio_convert_submit_input_buffer (GstBaseTransform *
base, gboolean is_discont, GstBuffer * input);
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
{
PROP_0,
PROP_DITHERING,
PROP_NOISE_SHAPING,
PROP_MIX_MATRIX,
};
#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);
/*** 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, PROP_DITHERING,
g_param_spec_enum ("dithering", "Dithering",
"Selects between different dithering methods.",
GST_TYPE_AUDIO_DITHER_METHOD, GST_AUDIO_DITHER_TPDF,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_NOISE_SHAPING,
g_param_spec_enum ("noise-shaping", "Noise shaping",
"Selects between different noise shaping methods.",
GST_TYPE_AUDIO_NOISE_SHAPING_METHOD, GST_AUDIO_NOISE_SHAPING_NONE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MIX_MATRIX,
gst_param_spec_array ("mix-matrix",
"Input/output channel matrix",
"Transformation matrix for input/output channels",
gst_param_spec_array ("matrix-rows", "rows", "rows",
g_param_spec_float ("matrix-cols", "cols", "cols",
-1, 1, 0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS),
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS),
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_static_pad_template (element_class,
&gst_audio_convert_src_template);
gst_element_class_add_static_pad_template (element_class,
&gst_audio_convert_sink_template);
gst_element_class_set_static_metadata (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 =
GST_DEBUG_FUNCPTR (gst_audio_convert_transform);
basetransform_class->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_convert_transform_ip);
basetransform_class->transform_meta =
GST_DEBUG_FUNCPTR (gst_audio_convert_transform_meta);
basetransform_class->submit_input_buffer =
GST_DEBUG_FUNCPTR (gst_audio_convert_submit_input_buffer);
basetransform_class->passthrough_on_same_caps = TRUE;
basetransform_class->transform_ip_on_passthrough = FALSE;
}
static void
gst_audio_convert_init (GstAudioConvert * this)
{
this->dither = GST_AUDIO_DITHER_TPDF;
this->ns = GST_AUDIO_NOISE_SHAPING_NONE;
g_value_init (&this->mix_matrix, GST_TYPE_ARRAY);
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);
if (this->convert) {
gst_audio_converter_free (this->convert);
this->convert = NULL;
}
g_value_unset (&this->mix_matrix);
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;
}
}
static gboolean
remove_format_from_structure (GstCapsFeatures * features,
GstStructure * structure, gpointer user_data G_GNUC_UNUSED)
{
gst_structure_remove_field (structure, "format");
return TRUE;
}
static gboolean
remove_channels_from_structure (GstCapsFeatures * features, GstStructure * s,
gpointer user_data)
{
guint64 mask;
gint channels;
GstAudioConvert *this = GST_AUDIO_CONVERT (user_data);
/* Only remove the channels and channel-mask for non-NONE layouts,
* or if a mix matrix was manually specified */
if (this->mix_matrix_was_set ||
!gst_structure_get (s, "channel-mask", GST_TYPE_BITMASK, &mask, NULL) ||
(mask != 0 || (gst_structure_get_int (s, "channels", &channels)
&& channels == 1))) {
gst_structure_remove_fields (s, "channel-mask", "channels", NULL);
}
return TRUE;
}
static gboolean
add_other_channels_to_structure (GstCapsFeatures * features, GstStructure * s,
gpointer user_data)
{
gint other_channels = GPOINTER_TO_INT (user_data);
gst_structure_set (s, "channels", G_TYPE_INT, other_channels, NULL);
return TRUE;
}
/* 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;
GstAudioConvert *this = GST_AUDIO_CONVERT (btrans);
tmp = gst_caps_copy (caps);
gst_caps_map_in_place (tmp, remove_format_from_structure, NULL);
gst_caps_map_in_place (tmp, remove_channels_from_structure, btrans);
/* We can infer the required input / output channels based on the
* matrix dimensions */
if (gst_value_array_get_size (&this->mix_matrix)) {
gint other_channels;
if (direction == GST_PAD_SRC) {
const GValue *first_row =
gst_value_array_get_value (&this->mix_matrix, 0);
other_channels = gst_value_array_get_size (first_row);
} else {
other_channels = gst_value_array_get_size (&this->mix_matrix);
}
gst_caps_map_in_place (tmp, add_other_channels_to_structure,
GINT_TO_POINTER (other_channels));
}
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;
}
/* Count the number of bits set
* Optimized for the common case, assuming that the number of channels
* (i.e. bits set) is small
*/
static gint
n_bits_set (guint64 x)
{
gint c;
for (c = 0; x; c++)
x &= x - 1;
return c;
}
/* Reduce the mask to the n_chans lowest set bits
*
* The algorithm clears the n_chans lowest set bits and subtracts the
* result from the original mask to get the desired mask.
* It is optimized for the common case where n_chans is a small
* number. In the worst case, however, it stops after 64 iterations.
*/
static guint64
find_suitable_mask (guint64 mask, gint n_chans)
{
guint64 x = mask;
for (; x && n_chans; n_chans--)
x &= x - 1;
g_assert (x || n_chans == 0);
/* assertion fails if mask contained less bits than n_chans
* or n_chans was < 0 */
return mask - x;
}
static void
gst_audio_convert_fixate_format (GstBaseTransform * base, GstStructure * ins,
GstStructure * outs)
{
const gchar *in_format;
const GValue *format;
const GstAudioFormatInfo *in_info, *out_info = NULL;
GstAudioFormatFlags in_flags, out_flags = 0;
gint in_depth, out_depth = -1;
gint i, len;
in_format = gst_structure_get_string (ins, "format");
if (!in_format)
return;
format = gst_structure_get_value (outs, "format");
/* should not happen */
if (format == NULL)
return;
/* nothing to fixate? */
if (!GST_VALUE_HOLDS_LIST (format))
return;
in_info =
gst_audio_format_get_info (gst_audio_format_from_string (in_format));
if (!in_info)
return;
in_flags = GST_AUDIO_FORMAT_INFO_FLAGS (in_info);
in_flags &= ~(GST_AUDIO_FORMAT_FLAG_UNPACK);
in_flags &= ~(GST_AUDIO_FORMAT_FLAG_SIGNED);
in_depth = GST_AUDIO_FORMAT_INFO_DEPTH (in_info);
len = gst_value_list_get_size (format);
for (i = 0; i < len; i++) {
const GstAudioFormatInfo *t_info;
GstAudioFormatFlags t_flags;
gboolean t_flags_better;
const GValue *val;
const gchar *fname;
gint t_depth;
val = gst_value_list_get_value (format, i);
if (!G_VALUE_HOLDS_STRING (val))
continue;
fname = g_value_get_string (val);
t_info = gst_audio_format_get_info (gst_audio_format_from_string (fname));
if (!t_info)
continue;
/* accept input format immediately */
if (strcmp (fname, in_format) == 0) {
out_info = t_info;
break;
}
t_flags = GST_AUDIO_FORMAT_INFO_FLAGS (t_info);
t_flags &= ~(GST_AUDIO_FORMAT_FLAG_UNPACK);
t_flags &= ~(GST_AUDIO_FORMAT_FLAG_SIGNED);
t_depth = GST_AUDIO_FORMAT_INFO_DEPTH (t_info);
/* Any output format is better than no output format at all */
if (!out_info) {
out_info = t_info;
out_depth = t_depth;
out_flags = t_flags;
continue;
}
t_flags_better = (t_flags == in_flags && out_flags != in_flags);
if (t_depth == in_depth && (out_depth != in_depth || t_flags_better)) {
/* Prefer to use the first format that has the same depth with the same
* flags, and if none with the same flags exist use the first other one
* that has the same depth */
out_info = t_info;
out_depth = t_depth;
out_flags = t_flags;
} else if (t_depth >= in_depth && (in_depth > out_depth
|| (out_depth >= in_depth && t_flags_better))) {
/* Otherwise use the first format that has a higher depth with the same flags,
* if none with the same flags exist use the first other one that has a higher
* depth */
out_info = t_info;
out_depth = t_depth;
out_flags = t_flags;
} else if ((t_depth > out_depth && out_depth < in_depth)
|| (t_flags_better && out_depth == t_depth)) {
/* Else get at least the one with the highest depth, ideally with the same flags */
out_info = t_info;
out_depth = t_depth;
out_flags = t_flags;
}
}
if (out_info)
gst_structure_set (outs, "format", G_TYPE_STRING,
GST_AUDIO_FORMAT_INFO_NAME (out_info), NULL);
}
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). For mono, no mask is required and the fallback mask is 0 */
if (out_chans > 1
&& (out_mask = gst_audio_channel_get_fallback_mask (out_chans))) {
GST_DEBUG_OBJECT (base, "using default channel layout as fallback");
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, out_mask, NULL);
} else if (out_chans > 1) {
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)) {
GstCaps *removed = gst_caps_copy (caps);
if (result)
gst_caps_unref (result);
gst_caps_map_in_place (removed, remove_format_from_structure, NULL);
result = gst_caps_intersect (othercaps, removed);
gst_caps_unref (removed);
if (gst_caps_is_empty (result)) {
if (result)
gst_caps_unref (result);
result = othercaps;
} else {
gst_caps_unref (othercaps);
}
} else {
gst_caps_unref (othercaps);
}
GST_DEBUG_OBJECT (base, "now fixating %" GST_PTR_FORMAT, result);
/* 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_audio_convert_fixate_format (base, ins, outs);
/* fixate remaining */
result = 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;
gboolean in_place;
GstStructure *config;
GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
if (this->convert) {
gst_audio_converter_free (this->convert);
this->convert = NULL;
}
if (!gst_audio_info_from_caps (&in_info, incaps))
goto invalid_in;
if (!gst_audio_info_from_caps (&out_info, outcaps))
goto invalid_out;
config = gst_structure_new ("GstAudioConverterConfig",
GST_AUDIO_CONVERTER_OPT_DITHER_METHOD, GST_TYPE_AUDIO_DITHER_METHOD,
this->dither,
GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD,
GST_TYPE_AUDIO_NOISE_SHAPING_METHOD, this->ns, NULL);
if (this->mix_matrix_was_set)
gst_structure_set_value (config, GST_AUDIO_CONVERTER_OPT_MIX_MATRIX,
&this->mix_matrix);
this->convert = gst_audio_converter_new (0, &in_info, &out_info, config);
if (this->convert == NULL)
goto no_converter;
in_place = gst_audio_converter_supports_inplace (this->convert);
gst_base_transform_set_in_place (base, in_place);
this->in_info = in_info;
this->out_info = out_info;
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 make converter");
return FALSE;
}
}
/* if called through gst_audio_convert_transform_ip() inbuf == outbuf */
static GstFlowReturn
gst_audio_convert_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstFlowReturn ret;
GstAudioConvert *this = GST_AUDIO_CONVERT (base);
GstMapInfo srcmap = { NULL, }, dstmap;
gint insize, outsize;
gboolean inbuf_writable;
GstAudioConverterFlags flags;
gsize samples;
/* get amount of samples to convert. */
samples = gst_buffer_get_size (inbuf) / this->in_info.bpf;
/* get in/output sizes, to see if the buffers we got are of correct
* sizes */
insize = samples * this->in_info.bpf;
outsize = samples * this->out_info.bpf;
if (insize == 0 || outsize == 0)
return GST_FLOW_OK;
gst_buffer_resize (outbuf, 0, outsize);
/* get src and dst data */
if (inbuf != outbuf) {
inbuf_writable = gst_buffer_is_writable (inbuf)
&& gst_buffer_n_memory (inbuf) == 1
&& gst_memory_is_writable (gst_buffer_peek_memory (inbuf, 0));
if (!gst_buffer_map (inbuf, &srcmap,
inbuf_writable ? GST_MAP_READWRITE : GST_MAP_READ))
goto inmap_error;
} else {
inbuf_writable = TRUE;
}
if (!gst_buffer_map (outbuf, &dstmap, GST_MAP_WRITE))
goto outmap_error;
/* check in and outsize */
if (inbuf != outbuf) {
if (srcmap.size < insize)
goto wrong_size;
}
if (dstmap.size < outsize)
goto wrong_size;
/* and convert the samples */
flags = 0;
if (inbuf_writable)
flags |= GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE;
if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
gpointer in[1] = { srcmap.data };
gpointer out[1] = { dstmap.data };
if (!gst_audio_converter_samples (this->convert, flags,
inbuf != outbuf ? in : out, samples, out, samples))
goto convert_error;
} else {
/* Create silence buffer */
gst_audio_format_fill_silence (this->out_info.finfo, dstmap.data, outsize);
}
ret = GST_FLOW_OK;
done:
gst_buffer_unmap (outbuf, &dstmap);
if (inbuf != outbuf)
gst_buffer_unmap (inbuf, &srcmap);
return ret;
/* ERRORS */
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;
}
inmap_error:
{
GST_ELEMENT_ERROR (this, STREAM, FORMAT,
(NULL), ("failed to map input buffer"));
return GST_FLOW_ERROR;
}
outmap_error:
{
GST_ELEMENT_ERROR (this, STREAM, FORMAT,
(NULL), ("failed to map output buffer"));
if (inbuf != outbuf)
gst_buffer_unmap (inbuf, &srcmap);
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_audio_convert_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
return gst_audio_convert_transform (base, buf, buf);
}
static gboolean
gst_audio_convert_transform_meta (GstBaseTransform * trans, GstBuffer * outbuf,
GstMeta * meta, GstBuffer * inbuf)
{
const GstMetaInfo *info = meta->info;
const gchar *const *tags;
tags = gst_meta_api_type_get_tags (info->api);
if (!tags || (g_strv_length ((gchar **) tags) == 1
&& gst_meta_api_type_has_tag (info->api,
g_quark_from_string (GST_META_TAG_AUDIO_STR))))
return TRUE;
return FALSE;
}
static GstFlowReturn
gst_audio_convert_submit_input_buffer (GstBaseTransform * base,
gboolean is_discont, GstBuffer * input)
{
GstAudioConvert *this = GST_AUDIO_CONVERT (base);
if (base->segment.format == GST_FORMAT_TIME) {
input =
gst_audio_buffer_clip (input, &base->segment, this->in_info.rate,
this->in_info.bpf);
if (!input)
return GST_FLOW_OK;
}
return GST_BASE_TRANSFORM_CLASS (parent_class)->submit_input_buffer (base,
is_discont, input);
}
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 PROP_DITHERING:
this->dither = g_value_get_enum (value);
break;
case PROP_NOISE_SHAPING:
this->ns = g_value_get_enum (value);
break;
case PROP_MIX_MATRIX:
if (!gst_value_array_get_size (value)) {
g_value_copy (value, &this->mix_matrix);
this->mix_matrix_was_set = TRUE;
} else {
const GValue *first_row = gst_value_array_get_value (value, 0);
if (gst_value_array_get_size (first_row)) {
if (gst_value_array_get_size (&this->mix_matrix))
g_value_unset (&this->mix_matrix);
g_value_copy (value, &this->mix_matrix);
this->mix_matrix_was_set = TRUE;
} else {
g_warning ("Empty mix matrix's first row");
}
}
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 PROP_DITHERING:
g_value_set_enum (value, this->dither);
break;
case PROP_NOISE_SHAPING:
g_value_set_enum (value, this->ns);
break;
case PROP_MIX_MATRIX:
if (this->mix_matrix_was_set)
g_value_copy (&this->mix_matrix, value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}