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gstreamer/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c

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/* GStreamer
* Copyright (C) 2005 Wim Taymans <wim at fluendo dot com>
* (C) 2015 Wim Taymans <wim.taymans@gmail.com>
*
* audioconverter.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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <math.h>
#include <string.h>
#include "audio-converter.h"
#include "gstaudiopack.h"
/**
* SECTION:gstaudioconverter
* @title: GstAudioConverter
* @short_description: Generic audio conversion
*
* This object is used to convert audio samples from one format to another.
* The object can perform conversion of:
*
* * audio format with optional dithering and noise shaping
*
* * audio samplerate
*
* * audio channels and channel layout
*
*/
#ifndef GST_DISABLE_GST_DEBUG
#define GST_CAT_DEFAULT ensure_debug_category()
static GstDebugCategory *
ensure_debug_category (void)
{
static gsize cat_gonce = 0;
if (g_once_init_enter (&cat_gonce)) {
gsize cat_done;
cat_done = (gsize) _gst_debug_category_new ("audio-converter", 0,
"audio-converter object");
g_once_init_leave (&cat_gonce, cat_done);
}
return (GstDebugCategory *) cat_gonce;
}
#else
#define ensure_debug_category() /* NOOP */
#endif /* GST_DISABLE_GST_DEBUG */
typedef struct _AudioChain AudioChain;
typedef void (*AudioConvertFunc) (gpointer dst, const gpointer src, gint count);
typedef gboolean (*AudioConvertSamplesFunc) (GstAudioConverter * convert,
GstAudioConverterFlags flags, gpointer in[], gsize in_frames,
gpointer out[], gsize out_frames);
typedef void (*AudioConvertEndianFunc) (gpointer dst, const gpointer src,
gint count);
/* int/int int/float float/int float/float
*
* unpack S32 S32 F64 F64
* convert S32->F64
* channel mix S32 F64 F64 F64
* convert F64->S32
* quantize S32 S32
* pack S32 F64 S32 F64
*
*
* interleave
* deinterleave
* resample
*/
struct _GstAudioConverter
{
GstAudioInfo in;
GstAudioInfo out;
GstStructure *config;
GstAudioConverterFlags flags;
GstAudioFormat current_format;
GstAudioLayout current_layout;
gint current_channels;
gboolean in_writable;
gpointer *in_data;
gsize in_frames;
gpointer *out_data;
gsize out_frames;
gboolean in_place; /* the conversion can be done in place; returned by gst_audio_converter_supports_inplace() */
gboolean passthrough;
/* unpack */
gboolean in_default;
gboolean unpack_ip;
/* convert in */
AudioConvertFunc convert_in;
/* channel mix */
gboolean mix_passthrough;
GstAudioChannelMixer *mix;
/* resample */
GstAudioResampler *resampler;
/* convert out */
AudioConvertFunc convert_out;
/* quant */
GstAudioQuantize *quant;
/* change layout */
GstAudioFormat chlayout_format;
GstAudioLayout chlayout_target;
gint chlayout_channels;
/* pack */
gboolean out_default;
AudioChain *chain_end; /* NULL for empty chain or points to the last element in the chain */
/* endian swap */
AudioConvertEndianFunc swap_endian;
AudioConvertSamplesFunc convert;
};
static GstAudioConverter *
gst_audio_converter_copy (GstAudioConverter * convert)
{
GstAudioConverter *res =
gst_audio_converter_new (convert->flags, &convert->in, &convert->out,
convert->config);
return res;
}
G_DEFINE_BOXED_TYPE (GstAudioConverter, gst_audio_converter,
(GBoxedCopyFunc) gst_audio_converter_copy,
(GBoxedFreeFunc) gst_audio_converter_free);
typedef gboolean (*AudioChainFunc) (AudioChain * chain, gpointer user_data);
typedef gpointer *(*AudioChainAllocFunc) (AudioChain * chain, gsize num_samples,
gpointer user_data);
struct _AudioChain
{
AudioChain *prev;
AudioChainFunc make_func;
gpointer make_func_data;
GDestroyNotify make_func_notify;
const GstAudioFormatInfo *finfo;
gint stride;
gint inc;
gint blocks;
gboolean pass_alloc;
gboolean allow_ip;
AudioChainAllocFunc alloc_func;
gpointer alloc_data;
gpointer *tmp;
gsize allocated_samples;
gpointer *samples;
gsize num_samples;
};
static AudioChain *
audio_chain_new (AudioChain * prev, GstAudioConverter * convert)
{
AudioChain *chain;
chain = g_slice_new0 (AudioChain);
chain->prev = prev;
if (convert->current_layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
chain->inc = 1;
chain->blocks = convert->current_channels;
} else {
chain->inc = convert->current_channels;
chain->blocks = 1;
}
chain->finfo = gst_audio_format_get_info (convert->current_format);
chain->stride = (chain->finfo->width * chain->inc) / 8;
return chain;
}
static void
audio_chain_set_make_func (AudioChain * chain,
AudioChainFunc make_func, gpointer user_data, GDestroyNotify notify)
{
chain->make_func = make_func;
chain->make_func_data = user_data;
chain->make_func_notify = notify;
}
static void
audio_chain_free (AudioChain * chain)
{
GST_LOG ("free chain %p", chain);
if (chain->make_func_notify)
chain->make_func_notify (chain->make_func_data);
g_free (chain->tmp);
g_slice_free (AudioChain, chain);
}
static gpointer *
audio_chain_alloc_samples (AudioChain * chain, gsize num_samples)
{
return chain->alloc_func (chain, num_samples, chain->alloc_data);
}
static void
audio_chain_set_samples (AudioChain * chain, gpointer * samples,
gsize num_samples)
{
GST_LOG ("set samples %p %" G_GSIZE_FORMAT, samples, num_samples);
chain->samples = samples;
chain->num_samples = num_samples;
}
static gpointer *
audio_chain_get_samples (AudioChain * chain, gsize * avail)
{
gpointer *res;
if (!chain->samples)
chain->make_func (chain, chain->make_func_data);
res = chain->samples;
*avail = chain->num_samples;
chain->samples = NULL;
return res;
}
static guint
get_opt_uint (GstAudioConverter * convert, const gchar * opt, guint def)
{
guint res;
if (!gst_structure_get_uint (convert->config, opt, &res))
res = def;
return res;
}
static gint
get_opt_enum (GstAudioConverter * convert, const gchar * opt, GType type,
gint def)
{
gint res;
if (!gst_structure_get_enum (convert->config, opt, type, &res))
res = def;
return res;
}
static const GValue *
get_opt_value (GstAudioConverter * convert, const gchar * opt)
{
return gst_structure_get_value (convert->config, opt);
}
#define DEFAULT_OPT_RESAMPLER_METHOD GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL
#define DEFAULT_OPT_DITHER_METHOD GST_AUDIO_DITHER_NONE
#define DEFAULT_OPT_DITHER_THRESHOLD 20
#define DEFAULT_OPT_NOISE_SHAPING_METHOD GST_AUDIO_NOISE_SHAPING_NONE
#define DEFAULT_OPT_QUANTIZATION 1
#define GET_OPT_RESAMPLER_METHOD(c) get_opt_enum(c, \
GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD, GST_TYPE_AUDIO_RESAMPLER_METHOD, \
DEFAULT_OPT_RESAMPLER_METHOD)
#define GET_OPT_DITHER_METHOD(c) get_opt_enum(c, \
GST_AUDIO_CONVERTER_OPT_DITHER_METHOD, GST_TYPE_AUDIO_DITHER_METHOD, \
DEFAULT_OPT_DITHER_METHOD)
#define GET_OPT_DITHER_THRESHOLD(c) get_opt_uint(c, \
GST_AUDIO_CONVERTER_OPT_DITHER_THRESHOLD, DEFAULT_OPT_DITHER_THRESHOLD)
#define GET_OPT_NOISE_SHAPING_METHOD(c) get_opt_enum(c, \
GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD, GST_TYPE_AUDIO_NOISE_SHAPING_METHOD, \
DEFAULT_OPT_NOISE_SHAPING_METHOD)
#define GET_OPT_QUANTIZATION(c) get_opt_uint(c, \
GST_AUDIO_CONVERTER_OPT_QUANTIZATION, DEFAULT_OPT_QUANTIZATION)
#define GET_OPT_MIX_MATRIX(c) get_opt_value(c, \
GST_AUDIO_CONVERTER_OPT_MIX_MATRIX)
static gboolean
copy_config (GQuark field_id, const GValue * value, gpointer user_data)
{
GstAudioConverter *convert = user_data;
gst_structure_id_set_value (convert->config, field_id, value);
return TRUE;
}
/**
* gst_audio_converter_update_config:
* @convert: a #GstAudioConverter
* @in_rate: input rate
* @out_rate: output rate
* @config: (transfer full) (allow-none): a #GstStructure or %NULL
*
* Set @in_rate, @out_rate and @config as extra configuration for @convert.
*
* @in_rate and @out_rate specify the new sample rates of input and output
* formats. A value of 0 leaves the sample rate unchanged.
*
* @config can be %NULL, in which case, the current configuration is not
* changed.
*
* If the parameters in @config can not be set exactly, this function returns
* %FALSE and will try to update as much state as possible. The new state can
* then be retrieved and refined with gst_audio_converter_get_config().
*
* Look at the `GST_AUDIO_CONVERTER_OPT_*` fields to check valid configuration
* option and values.
*
* Returns: %TRUE when the new parameters could be set
*/
gboolean
gst_audio_converter_update_config (GstAudioConverter * convert,
gint in_rate, gint out_rate, GstStructure * config)
{
g_return_val_if_fail (convert != NULL, FALSE);
g_return_val_if_fail ((in_rate == 0 && out_rate == 0) ||
convert->flags & GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE, FALSE);
GST_LOG ("new rate %d -> %d", in_rate, out_rate);
if (in_rate <= 0)
in_rate = convert->in.rate;
if (out_rate <= 0)
out_rate = convert->out.rate;
convert->in.rate = in_rate;
convert->out.rate = out_rate;
if (convert->resampler)
gst_audio_resampler_update (convert->resampler, in_rate, out_rate, config);
if (config) {
gst_structure_foreach (config, copy_config, convert);
gst_structure_free (config);
}
return TRUE;
}
/**
* gst_audio_converter_get_config:
* @convert: a #GstAudioConverter
* @in_rate: (out) (optional): result input rate
* @out_rate: (out) (optional): result output rate
*
* Get the current configuration of @convert.
*
* Returns: (transfer none):
* a #GstStructure that remains valid for as long as @convert is valid
* or until gst_audio_converter_update_config() is called.
*/
const GstStructure *
gst_audio_converter_get_config (GstAudioConverter * convert,
gint * in_rate, gint * out_rate)
{
g_return_val_if_fail (convert != NULL, NULL);
if (in_rate)
*in_rate = convert->in.rate;
if (out_rate)
*out_rate = convert->out.rate;
return convert->config;
}
static gpointer *
get_output_samples (AudioChain * chain, gsize num_samples, gpointer user_data)
{
GstAudioConverter *convert = user_data;
GST_LOG ("output samples %p %" G_GSIZE_FORMAT, convert->out_data,
num_samples);
return convert->out_data;
}
#define MEM_ALIGN(m,a) ((gint8 *)((guintptr)((gint8 *)(m) + ((a)-1)) & ~((a)-1)))
#define ALIGN 16
static gpointer *
get_temp_samples (AudioChain * chain, gsize num_samples, gpointer user_data)
{
if (num_samples > chain->allocated_samples) {
gint i;
gint8 *s;
gsize stride = GST_ROUND_UP_N (num_samples * chain->stride, ALIGN);
/* first part contains the pointers, second part the data, add some extra bytes
* for alignment */
gsize needed = (stride + sizeof (gpointer)) * chain->blocks + ALIGN - 1;
GST_DEBUG ("alloc samples %d %" G_GSIZE_FORMAT " %" G_GSIZE_FORMAT,
chain->stride, num_samples, needed);
chain->tmp = g_realloc (chain->tmp, needed);
chain->allocated_samples = num_samples;
/* pointer to the data, make sure it's 16 bytes aligned */
s = MEM_ALIGN (&chain->tmp[chain->blocks], ALIGN);
/* set up the pointers */
for (i = 0; i < chain->blocks; i++)
chain->tmp[i] = s + i * stride;
}
GST_LOG ("temp samples %p %" G_GSIZE_FORMAT, chain->tmp, num_samples);
return chain->tmp;
}
static gboolean
do_unpack (AudioChain * chain, gpointer user_data)
{
GstAudioConverter *convert = user_data;
gsize num_samples;
gpointer *tmp;
gboolean in_writable;
in_writable = convert->in_writable;
num_samples = convert->in_frames;
if (!chain->allow_ip || !in_writable || !convert->in_default) {
gint i;
if (in_writable && chain->allow_ip) {
tmp = convert->in_data;
GST_LOG ("unpack in-place %p, %" G_GSIZE_FORMAT, tmp, num_samples);
} else {
tmp = audio_chain_alloc_samples (chain, num_samples);
GST_LOG ("unpack to tmp %p, %" G_GSIZE_FORMAT, tmp, num_samples);
}
if (convert->in_data) {
for (i = 0; i < chain->blocks; i++) {
if (convert->in_default) {
GST_LOG ("copy %p, %p, %" G_GSIZE_FORMAT, tmp[i], convert->in_data[i],
num_samples);
memcpy (tmp[i], convert->in_data[i], num_samples * chain->stride);
} else {
GST_LOG ("unpack %p, %p, %" G_GSIZE_FORMAT, tmp[i],
convert->in_data[i], num_samples);
convert->in.finfo->unpack_func (convert->in.finfo,
GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE, tmp[i], convert->in_data[i],
num_samples * chain->inc);
}
}
} else {
for (i = 0; i < chain->blocks; i++) {
gst_audio_format_info_fill_silence (chain->finfo, tmp[i],
num_samples * chain->inc);
}
}
} else {
tmp = convert->in_data;
GST_LOG ("get in samples %p", tmp);
}
audio_chain_set_samples (chain, tmp, num_samples);
return TRUE;
}
static gboolean
do_convert_in (AudioChain * chain, gpointer user_data)
{
gsize num_samples;
GstAudioConverter *convert = user_data;
gpointer *in, *out;
gint i;
in = audio_chain_get_samples (chain->prev, &num_samples);
out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples));
GST_LOG ("convert in %p, %p, %" G_GSIZE_FORMAT, in, out, num_samples);
for (i = 0; i < chain->blocks; i++)
convert->convert_in (out[i], in[i], num_samples * chain->inc);
audio_chain_set_samples (chain, out, num_samples);
return TRUE;
}
static gboolean
do_mix (AudioChain * chain, gpointer user_data)
{
gsize num_samples;
GstAudioConverter *convert = user_data;
gpointer *in, *out;
in = audio_chain_get_samples (chain->prev, &num_samples);
out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples));
GST_LOG ("mix %p, %p, %" G_GSIZE_FORMAT, in, out, num_samples);
gst_audio_channel_mixer_samples (convert->mix, in, out, num_samples);
audio_chain_set_samples (chain, out, num_samples);
return TRUE;
}
static gboolean
do_resample (AudioChain * chain, gpointer user_data)
{
GstAudioConverter *convert = user_data;
gpointer *in, *out;
gsize in_frames, out_frames;
in = audio_chain_get_samples (chain->prev, &in_frames);
out_frames = convert->out_frames;
out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, out_frames));
GST_LOG ("resample %p %p,%" G_GSIZE_FORMAT " %" G_GSIZE_FORMAT, in,
out, in_frames, out_frames);
gst_audio_resampler_resample (convert->resampler, in, in_frames, out,
out_frames);
audio_chain_set_samples (chain, out, out_frames);
return TRUE;
}
static gboolean
do_convert_out (AudioChain * chain, gpointer user_data)
{
GstAudioConverter *convert = user_data;
gsize num_samples;
gpointer *in, *out;
gint i;
in = audio_chain_get_samples (chain->prev, &num_samples);
out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples));
GST_LOG ("convert out %p, %p %" G_GSIZE_FORMAT, in, out, num_samples);
for (i = 0; i < chain->blocks; i++)
convert->convert_out (out[i], in[i], num_samples * chain->inc);
audio_chain_set_samples (chain, out, num_samples);
return TRUE;
}
static gboolean
do_quantize (AudioChain * chain, gpointer user_data)
{
GstAudioConverter *convert = user_data;
gsize num_samples;
gpointer *in, *out;
in = audio_chain_get_samples (chain->prev, &num_samples);
out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples));
GST_LOG ("quantize %p, %p %" G_GSIZE_FORMAT, in, out, num_samples);
if (in && out)
gst_audio_quantize_samples (convert->quant, in, out, num_samples);
audio_chain_set_samples (chain, out, num_samples);
return TRUE;
}
#define MAKE_INTERLEAVE_FUNC(type) \
static inline void \
interleave_##type (const type * in[], type * out[], \
gsize num_samples, gint channels) \
{ \
gsize s; \
gint c; \
for (s = 0; s < num_samples; s++) { \
for (c = 0; c < channels; c++) { \
out[0][s * channels + c] = in[c][s]; \
} \
} \
}
#define MAKE_DEINTERLEAVE_FUNC(type) \
static inline void \
deinterleave_##type (const type * in[], type * out[], \
gsize num_samples, gint channels) \
{ \
gsize s; \
gint c; \
for (s = 0; s < num_samples; s++) { \
for (c = 0; c < channels; c++) { \
out[c][s] = in[0][s * channels + c]; \
} \
} \
}
MAKE_INTERLEAVE_FUNC (gint16);
MAKE_INTERLEAVE_FUNC (gint32);
MAKE_INTERLEAVE_FUNC (gfloat);
MAKE_INTERLEAVE_FUNC (gdouble);
MAKE_DEINTERLEAVE_FUNC (gint16);
MAKE_DEINTERLEAVE_FUNC (gint32);
MAKE_DEINTERLEAVE_FUNC (gfloat);
MAKE_DEINTERLEAVE_FUNC (gdouble);
static gboolean
do_change_layout (AudioChain * chain, gpointer user_data)
{
GstAudioConverter *convert = user_data;
GstAudioFormat format = convert->chlayout_format;
GstAudioLayout out_layout = convert->chlayout_target;
gint channels = convert->chlayout_channels;
gsize num_samples;
gpointer *in, *out;
in = audio_chain_get_samples (chain->prev, &num_samples);
out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples));
if (out_layout == GST_AUDIO_LAYOUT_INTERLEAVED) {
/* interleave */
GST_LOG ("interleaving %p, %p %" G_GSIZE_FORMAT, in, out, num_samples);
switch (format) {
case GST_AUDIO_FORMAT_S16:
interleave_gint16 ((const gint16 **) in, (gint16 **) out,
num_samples, channels);
break;
case GST_AUDIO_FORMAT_S32:
interleave_gint32 ((const gint32 **) in, (gint32 **) out,
num_samples, channels);
break;
case GST_AUDIO_FORMAT_F32:
interleave_gfloat ((const gfloat **) in, (gfloat **) out,
num_samples, channels);
break;
case GST_AUDIO_FORMAT_F64:
interleave_gdouble ((const gdouble **) in, (gdouble **) out,
num_samples, channels);
break;
default:
g_assert_not_reached ();
break;
}
} else {
/* deinterleave */
GST_LOG ("deinterleaving %p, %p %" G_GSIZE_FORMAT, in, out, num_samples);
switch (format) {
case GST_AUDIO_FORMAT_S16:
deinterleave_gint16 ((const gint16 **) in, (gint16 **) out,
num_samples, channels);
break;
case GST_AUDIO_FORMAT_S32:
deinterleave_gint32 ((const gint32 **) in, (gint32 **) out,
num_samples, channels);
break;
case GST_AUDIO_FORMAT_F32:
deinterleave_gfloat ((const gfloat **) in, (gfloat **) out,
num_samples, channels);
break;
case GST_AUDIO_FORMAT_F64:
deinterleave_gdouble ((const gdouble **) in, (gdouble **) out,
num_samples, channels);
break;
default:
g_assert_not_reached ();
break;
}
}
audio_chain_set_samples (chain, out, num_samples);
return TRUE;
}
static gboolean
is_intermediate_format (GstAudioFormat format)
{
return (format == GST_AUDIO_FORMAT_S16 ||
format == GST_AUDIO_FORMAT_S32 ||
format == GST_AUDIO_FORMAT_F32 || format == GST_AUDIO_FORMAT_F64);
}
static AudioChain *
chain_unpack (GstAudioConverter * convert)
{
AudioChain *prev;
GstAudioInfo *in = &convert->in;
GstAudioInfo *out = &convert->out;
gboolean same_format;
same_format = in->finfo->format == out->finfo->format;
/* do not unpack if we have the same input format as the output format
* and it is a possible intermediate format */
if (same_format && is_intermediate_format (in->finfo->format)) {
convert->current_format = in->finfo->format;
} else {
convert->current_format = in->finfo->unpack_format;
}
convert->current_layout = in->layout;
convert->current_channels = in->channels;
convert->in_default = convert->current_format == in->finfo->format;
GST_INFO ("unpack format %s to %s",
gst_audio_format_to_string (in->finfo->format),
gst_audio_format_to_string (convert->current_format));
prev = audio_chain_new (NULL, convert);
prev->allow_ip = prev->finfo->width <= in->finfo->width;
prev->pass_alloc = FALSE;
audio_chain_set_make_func (prev, do_unpack, convert, NULL);
return prev;
}
static AudioChain *
chain_convert_in (GstAudioConverter * convert, AudioChain * prev)
{
gboolean in_int, out_int;
GstAudioInfo *in = &convert->in;
GstAudioInfo *out = &convert->out;
in_int = GST_AUDIO_FORMAT_INFO_IS_INTEGER (in->finfo);
out_int = GST_AUDIO_FORMAT_INFO_IS_INTEGER (out->finfo);
if (in_int && !out_int) {
GST_INFO ("convert S32 to F64");
convert->convert_in = (AudioConvertFunc) audio_orc_s32_to_double;
convert->current_format = GST_AUDIO_FORMAT_F64;
prev = audio_chain_new (prev, convert);
prev->allow_ip = FALSE;
prev->pass_alloc = FALSE;
audio_chain_set_make_func (prev, do_convert_in, convert, NULL);
}
return prev;
}
static gboolean
check_mix_matrix (guint in_channels, guint out_channels, const GValue * value)
{
guint i, j;
/* audio-channel-mixer will generate an identity matrix */
if (gst_value_array_get_size (value) == 0)
return TRUE;
if (gst_value_array_get_size (value) != out_channels) {
GST_ERROR ("Invalid mix matrix size, should be %d", out_channels);
goto fail;
}
for (j = 0; j < out_channels; j++) {
const GValue *row = gst_value_array_get_value (value, j);
if (gst_value_array_get_size (row) != in_channels) {
GST_ERROR ("Invalid mix matrix row size, should be %d", in_channels);
goto fail;
}
for (i = 0; i < in_channels; i++) {
const GValue *itm;
itm = gst_value_array_get_value (row, i);
if (!G_VALUE_HOLDS_FLOAT (itm)) {
GST_ERROR ("Invalid mix matrix element type, should be float");
goto fail;
}
}
}
return TRUE;
fail:
return FALSE;
}
static gfloat **
mix_matrix_from_g_value (guint in_channels, guint out_channels,
const GValue * value)
{
guint i, j;
gfloat **matrix = g_new (gfloat *, in_channels);
for (i = 0; i < in_channels; i++)
matrix[i] = g_new (gfloat, out_channels);
for (j = 0; j < out_channels; j++) {
const GValue *row = gst_value_array_get_value (value, j);
for (i = 0; i < in_channels; i++) {
const GValue *itm;
gfloat coefficient;
itm = gst_value_array_get_value (row, i);
coefficient = g_value_get_float (itm);
matrix[i][j] = coefficient;
}
}
return matrix;
}
static AudioChain *
chain_mix (GstAudioConverter * convert, AudioChain * prev)
{
GstAudioInfo *in = &convert->in;
GstAudioInfo *out = &convert->out;
GstAudioFormat format = convert->current_format;
const GValue *opt_matrix = GET_OPT_MIX_MATRIX (convert);
GstAudioChannelMixerFlags flags = 0;
convert->current_channels = out->channels;
/* keep the input layout */
if (convert->current_layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
flags |= GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_IN;
flags |= GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_OUT;
}
if (opt_matrix) {
gfloat **matrix = NULL;
if (gst_value_array_get_size (opt_matrix))
matrix =
mix_matrix_from_g_value (in->channels, out->channels, opt_matrix);
convert->mix =
gst_audio_channel_mixer_new_with_matrix (flags, format, in->channels,
out->channels, matrix);
} else {
flags |=
GST_AUDIO_INFO_IS_UNPOSITIONED (in) ?
GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_IN : 0;
flags |=
GST_AUDIO_INFO_IS_UNPOSITIONED (out) ?
GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_OUT : 0;
convert->mix =
gst_audio_channel_mixer_new (flags, format, in->channels, in->position,
out->channels, out->position);
}
convert->mix_passthrough =
gst_audio_channel_mixer_is_passthrough (convert->mix);
GST_INFO ("mix format %s, passthrough %d, in_channels %d, out_channels %d",
gst_audio_format_to_string (format), convert->mix_passthrough,
in->channels, out->channels);
if (!convert->mix_passthrough) {
prev = audio_chain_new (prev, convert);
prev->allow_ip = FALSE;
prev->pass_alloc = FALSE;
audio_chain_set_make_func (prev, do_mix, convert, NULL);
}
return prev;
}
static AudioChain *
chain_resample (GstAudioConverter * convert, AudioChain * prev)
{
GstAudioInfo *in = &convert->in;
GstAudioInfo *out = &convert->out;
GstAudioResamplerMethod method;
GstAudioResamplerFlags flags;
GstAudioFormat format = convert->current_format;
gint channels = convert->current_channels;
gboolean variable_rate;
variable_rate = convert->flags & GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE;
if (in->rate != out->rate || variable_rate) {
method = GET_OPT_RESAMPLER_METHOD (convert);
flags = 0;
if (convert->current_layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
flags |= GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN;
}
/* if the resampler is activated, it is optimal to change layout here */
if (out->layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
flags |= GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT;
}
convert->current_layout = out->layout;
if (variable_rate)
flags |= GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE;
convert->resampler =
gst_audio_resampler_new (method, flags, format, channels, in->rate,
out->rate, convert->config);
prev = audio_chain_new (prev, convert);
prev->allow_ip = FALSE;
prev->pass_alloc = FALSE;
audio_chain_set_make_func (prev, do_resample, convert, NULL);
}
return prev;
}
static AudioChain *
chain_convert_out (GstAudioConverter * convert, AudioChain * prev)
{
gboolean in_int, out_int;
GstAudioInfo *in = &convert->in;
GstAudioInfo *out = &convert->out;
in_int = GST_AUDIO_FORMAT_INFO_IS_INTEGER (in->finfo);
out_int = GST_AUDIO_FORMAT_INFO_IS_INTEGER (out->finfo);
if (!in_int && out_int) {
convert->convert_out = (AudioConvertFunc) audio_orc_double_to_s32;
convert->current_format = GST_AUDIO_FORMAT_S32;
GST_INFO ("convert F64 to S32");
prev = audio_chain_new (prev, convert);
prev->allow_ip = TRUE;
prev->pass_alloc = FALSE;
audio_chain_set_make_func (prev, do_convert_out, convert, NULL);
}
return prev;
}
static AudioChain *
chain_quantize (GstAudioConverter * convert, AudioChain * prev)
{
const GstAudioFormatInfo *cur_finfo;
GstAudioInfo *out = &convert->out;
gint in_depth, out_depth;
gboolean in_int, out_int;
GstAudioDitherMethod dither;
guint dither_threshold;
GstAudioNoiseShapingMethod ns;
dither = GET_OPT_DITHER_METHOD (convert);
dither_threshold = GET_OPT_DITHER_THRESHOLD (convert);
ns = GET_OPT_NOISE_SHAPING_METHOD (convert);
cur_finfo = gst_audio_format_get_info (convert->current_format);
in_depth = GST_AUDIO_FORMAT_INFO_DEPTH (cur_finfo);
out_depth = GST_AUDIO_FORMAT_INFO_DEPTH (out->finfo);
GST_INFO ("depth in %d, out %d", in_depth, out_depth);
in_int = GST_AUDIO_FORMAT_INFO_IS_INTEGER (cur_finfo);
out_int = GST_AUDIO_FORMAT_INFO_IS_INTEGER (out->finfo);
/* Don't dither or apply noise shaping if target depth is bigger than 20 bits
* as DA converters only can do a SNR up to 20 bits in reality.
* Also don't dither or apply noise shaping if target depth is larger than
* source depth. */
if (out_depth > dither_threshold || (in_int && out_depth >= in_depth)) {
dither = GST_AUDIO_DITHER_NONE;
ns = GST_AUDIO_NOISE_SHAPING_NONE;
GST_INFO ("using no dither and noise shaping");
} else {
GST_INFO ("using dither %d and noise shaping %d", dither, ns);
/* Use simple error feedback when output sample rate is smaller than
* 32000 as the other methods might move the noise to audible ranges */
if (ns > GST_AUDIO_NOISE_SHAPING_ERROR_FEEDBACK && out->rate < 32000)
ns = GST_AUDIO_NOISE_SHAPING_ERROR_FEEDBACK;
}
/* we still want to run the quantization step when reducing bits to get
* the rounding correct */
if (out_int && out_depth < 32
&& convert->current_format == GST_AUDIO_FORMAT_S32) {
GST_INFO ("quantize to %d bits, dither %d, ns %d", out_depth, dither, ns);
convert->quant =
gst_audio_quantize_new (dither, ns, 0, convert->current_format,
out->channels, 1U << (32 - out_depth));
prev = audio_chain_new (prev, convert);
prev->allow_ip = TRUE;
prev->pass_alloc = TRUE;
audio_chain_set_make_func (prev, do_quantize, convert, NULL);
}
return prev;
}
static AudioChain *
chain_change_layout (GstAudioConverter * convert, AudioChain * prev)
{
GstAudioInfo *out = &convert->out;
if (convert->current_layout != out->layout) {
convert->current_layout = out->layout;
/* if there is only 1 channel, layouts are identical */
if (convert->current_channels > 1) {
convert->chlayout_target = convert->current_layout;
convert->chlayout_format = convert->current_format;
convert->chlayout_channels = convert->current_channels;
prev = audio_chain_new (prev, convert);
prev->allow_ip = FALSE;
prev->pass_alloc = FALSE;
audio_chain_set_make_func (prev, do_change_layout, convert, NULL);
}
}
return prev;
}
static AudioChain *
chain_pack (GstAudioConverter * convert, AudioChain * prev)
{
GstAudioInfo *out = &convert->out;
GstAudioFormat format = convert->current_format;
convert->current_format = out->finfo->format;
convert->out_default = format == out->finfo->format;
GST_INFO ("pack format %s to %s", gst_audio_format_to_string (format),
gst_audio_format_to_string (out->finfo->format));
return prev;
}
static void
setup_allocators (GstAudioConverter * convert)
{
AudioChain *chain;
AudioChainAllocFunc alloc_func;
gboolean allow_ip;
/* start with using dest if we can directly write into it */
if (convert->out_default) {
alloc_func = get_output_samples;
allow_ip = FALSE;
} else {
alloc_func = get_temp_samples;
allow_ip = TRUE;
}
/* now walk backwards, we try to write into the dest samples directly
* and keep track if the source needs to be writable */
for (chain = convert->chain_end; chain; chain = chain->prev) {
chain->alloc_func = alloc_func;
chain->alloc_data = convert;
chain->allow_ip = allow_ip && chain->allow_ip;
GST_LOG ("chain %p: %d %d", chain, allow_ip, chain->allow_ip);
if (!chain->pass_alloc) {
/* can't pass allocator, make new temp line allocator */
alloc_func = get_temp_samples;
allow_ip = TRUE;
}
}
}
static gboolean
converter_passthrough (GstAudioConverter * convert,
GstAudioConverterFlags flags, gpointer in[], gsize in_frames,
gpointer out[], gsize out_frames)
{
gint i;
AudioChain *chain;
gsize samples;
/* in-place passthrough -> do nothing */
if (in == out) {
g_assert (convert->in_place);
return TRUE;
}
chain = convert->chain_end;
samples = in_frames * chain->inc;
GST_LOG ("passthrough: %" G_GSIZE_FORMAT " / %" G_GSIZE_FORMAT " samples",
in_frames, samples);
if (in) {
gsize bytes;
bytes = samples * (convert->in.bpf / convert->in.channels);
for (i = 0; i < chain->blocks; i++) {
if (out[i] == in[i]) {
g_assert (convert->in_place);
continue;
}
memcpy (out[i], in[i], bytes);
}
} else {
for (i = 0; i < chain->blocks; i++)
gst_audio_format_info_fill_silence (convert->in.finfo, out[i], samples);
}
return TRUE;
}
/* perform LE<->BE conversion on a block of @count 16-bit samples
* dst may equal src for in-place conversion
*/
static void
converter_swap_endian_16 (gpointer dst, const gpointer src, gint count)
{
guint16 *out = dst;
const guint16 *in = src;
gint i;
for (i = 0; i < count; i++)
out[i] = GUINT16_SWAP_LE_BE (in[i]);
}
/* perform LE<->BE conversion on a block of @count 24-bit samples
* dst may equal src for in-place conversion
*
* naive algorithm, which performs better with -O3 and worse with -O2
* than the commented out optimized algorithm below
*/
static void
converter_swap_endian_24 (gpointer dst, const gpointer src, gint count)
{
guint8 *out = dst;
const guint8 *in = src;
gint i;
count *= 3;
for (i = 0; i < count; i += 3) {
guint8 x = in[i + 0];
out[i + 0] = in[i + 2];
out[i + 1] = in[i + 1];
out[i + 2] = x;
}
}
/* the below code performs better with -O2 but worse with -O3 */
#if 0
/* perform LE<->BE conversion on a block of @count 24-bit samples
* dst may equal src for in-place conversion
*
* assumes that dst and src are 32-bit aligned
*/
static void
converter_swap_endian_24 (gpointer dst, const gpointer src, gint count)
{
guint32 *out = dst;
const guint32 *in = src;
guint8 *out8;
const guint8 *in8;
gint i;
/* first convert 24-bit samples in multiples of 4 reading 3x 32-bits in one cycle
*
* input: A1 B1 C1 A2 , B2 C2 A3 B3 , C3 A4 B4 C4
* 32-bit endian swap: A2 C1 B1 A1 , B3 A3 C2 B2 , C4 B4 A4 C3
* <-- x --> <-- y --> , <-- z -->
*
* desired output: C1 B1 A1 C2 , B2 A2 C3 B3 , A3 C4 B4 A4
*/
for (i = 0; i < count / 4; i++, in += 3, out += 3) {
guint32 x, y, z;
x = GUINT32_SWAP_LE_BE (in[0]);
y = GUINT32_SWAP_LE_BE (in[1]);
z = GUINT32_SWAP_LE_BE (in[2]);
#if G_BYTE_ORDER == G_BIG_ENDIAN
out[0] = (x << 8) + ((y >> 8) & 0xff);
out[1] = (in[1] & 0xff0000ff) + ((x >> 8) & 0xff0000) + ((z << 8) & 0xff00);
out[2] = (z >> 8) + ((y << 8) & 0xff000000);
#else
out[0] = (x >> 8) + ((y << 8) & 0xff000000);
out[1] = (in[1] & 0xff0000ff) + ((x << 8) & 0xff00) + ((z >> 8) & 0xff0000);
out[2] = (z << 8) + ((y >> 8) & 0xff);
#endif
}
/* convert the remainder less efficiently */
for (out8 = (guint8 *) out, in8 = (const guint8 *) in, i = 0; i < (count & 3);
i++) {
guint8 x = in8[i + 0];
out8[i + 0] = in8[i + 2];
out8[i + 1] = in8[i + 1];
out8[i + 2] = x;
}
}
#endif
/* perform LE<->BE conversion on a block of @count 32-bit samples
* dst may equal src for in-place conversion
*/
static void
converter_swap_endian_32 (gpointer dst, const gpointer src, gint count)
{
guint32 *out = dst;
const guint32 *in = src;
gint i;
for (i = 0; i < count; i++)
out[i] = GUINT32_SWAP_LE_BE (in[i]);
}
/* perform LE<->BE conversion on a block of @count 64-bit samples
* dst may equal src for in-place conversion
*/
static void
converter_swap_endian_64 (gpointer dst, const gpointer src, gint count)
{
guint64 *out = dst;
const guint64 *in = src;
gint i;
for (i = 0; i < count; i++)
out[i] = GUINT64_SWAP_LE_BE (in[i]);
}
/* the worker function to perform endian-conversion only
* assuming finfo and foutinfo have the same depth
*/
static gboolean
converter_endian (GstAudioConverter * convert,
GstAudioConverterFlags flags, gpointer in[], gsize in_frames,
gpointer out[], gsize out_frames)
{
gint i;
AudioChain *chain;
gsize samples;
chain = convert->chain_end;
samples = in_frames * chain->inc;
GST_LOG ("convert endian: %" G_GSIZE_FORMAT " / %" G_GSIZE_FORMAT " samples",
in_frames, samples);
if (in) {
for (i = 0; i < chain->blocks; i++)
convert->swap_endian (out[i], in[i], samples);
} else {
for (i = 0; i < chain->blocks; i++)
gst_audio_format_info_fill_silence (convert->in.finfo, out[i], samples);
}
return TRUE;
}
static gboolean
converter_generic (GstAudioConverter * convert,
GstAudioConverterFlags flags, gpointer in[], gsize in_frames,
gpointer out[], gsize out_frames)
{
AudioChain *chain;
gpointer *tmp;
gint i;
gsize produced;
chain = convert->chain_end;
convert->in_writable = flags & GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE;
convert->in_data = in;
convert->in_frames = in_frames;
convert->out_data = out;
convert->out_frames = out_frames;
/* get frames to pack */
tmp = audio_chain_get_samples (chain, &produced);
if (!convert->out_default && tmp && out) {
GST_LOG ("pack %p, %p %" G_GSIZE_FORMAT, tmp, out, produced);
/* and pack if needed */
for (i = 0; i < chain->blocks; i++)
convert->out.finfo->pack_func (convert->out.finfo, 0, tmp[i], out[i],
produced * chain->inc);
}
return TRUE;
}
static gboolean
converter_resample (GstAudioConverter * convert,
GstAudioConverterFlags flags, gpointer in[], gsize in_frames,
gpointer out[], gsize out_frames)
{
gst_audio_resampler_resample (convert->resampler, in, in_frames, out,
out_frames);
return TRUE;
}
#define GST_AUDIO_FORMAT_IS_ENDIAN_CONVERSION(info1, info2) \
( \
!(((info1)->flags ^ (info2)->flags) & (~GST_AUDIO_FORMAT_FLAG_UNPACK)) && \
(info1)->endianness != (info2)->endianness && \
(info1)->width == (info2)->width && \
(info1)->depth == (info2)->depth \
)
/**
* gst_audio_converter_new:
* @flags: extra #GstAudioConverterFlags
* @in_info: a source #GstAudioInfo
* @out_info: a destination #GstAudioInfo
* @config: (transfer full) (nullable): a #GstStructure with configuration options
*
* Create a new #GstAudioConverter that is able to convert between @in and @out
* audio formats.
*
* @config contains extra configuration options, see `GST_AUDIO_CONVERTER_OPT_*`
* parameters for details about the options and values.
*
* Returns: (nullable): a #GstAudioConverter or %NULL if conversion is not possible.
*/
GstAudioConverter *
gst_audio_converter_new (GstAudioConverterFlags flags, GstAudioInfo * in_info,
GstAudioInfo * out_info, GstStructure * config)
{
GstAudioConverter *convert;
AudioChain *prev;
const GValue *opt_matrix = NULL;
g_return_val_if_fail (in_info != NULL, FALSE);
g_return_val_if_fail (out_info != NULL, FALSE);
if (config)
opt_matrix =
gst_structure_get_value (config, GST_AUDIO_CONVERTER_OPT_MIX_MATRIX);
if (opt_matrix
&& !check_mix_matrix (in_info->channels, out_info->channels, opt_matrix))
goto invalid_mix_matrix;
if ((GST_AUDIO_INFO_CHANNELS (in_info) != GST_AUDIO_INFO_CHANNELS (out_info))
&& (GST_AUDIO_INFO_IS_UNPOSITIONED (in_info)
|| GST_AUDIO_INFO_IS_UNPOSITIONED (out_info))
&& !opt_matrix)
goto unpositioned;
convert = g_slice_new0 (GstAudioConverter);
convert->flags = flags;
convert->in = *in_info;
convert->out = *out_info;
/* default config */
convert->config = gst_structure_new_empty ("GstAudioConverter");
if (config)
gst_audio_converter_update_config (convert, 0, 0, config);
GST_INFO ("unitsizes: %d -> %d", in_info->bpf, out_info->bpf);
/* step 1, unpack */
prev = chain_unpack (convert);
/* step 2, optional convert from S32 to F64 for channel mix */
prev = chain_convert_in (convert, prev);
/* step 3, channel mix */
prev = chain_mix (convert, prev);
/* step 4, resample */
prev = chain_resample (convert, prev);
/* step 5, optional convert for quantize */
prev = chain_convert_out (convert, prev);
/* step 6, optional quantize */
prev = chain_quantize (convert, prev);
/* step 7, change layout */
prev = chain_change_layout (convert, prev);
/* step 8, pack */
convert->chain_end = chain_pack (convert, prev);
convert->convert = converter_generic;
convert->in_place = FALSE;
convert->passthrough = FALSE;
/* optimize */
if (convert->mix_passthrough) {
if (out_info->finfo->format == in_info->finfo->format) {
if (convert->resampler == NULL) {
if (out_info->layout == in_info->layout) {
GST_INFO ("same formats, same layout, no resampler and "
"passthrough mixing -> passthrough");
convert->convert = converter_passthrough;
convert->in_place = TRUE;
convert->passthrough = TRUE;
}
} else {
if (is_intermediate_format (in_info->finfo->format)) {
GST_INFO ("same formats, and passthrough mixing -> only resampling");
convert->convert = converter_resample;
}
}
} else if (GST_AUDIO_FORMAT_IS_ENDIAN_CONVERSION (out_info->finfo,
in_info->finfo)) {
if (convert->resampler == NULL && out_info->layout == in_info->layout) {
GST_INFO ("no resampler, passthrough mixing -> only endian conversion");
convert->convert = converter_endian;
convert->in_place = TRUE;
switch (GST_AUDIO_INFO_WIDTH (in_info)) {
case 16:
GST_DEBUG ("initializing 16-bit endian conversion");
convert->swap_endian = converter_swap_endian_16;
break;
case 24:
GST_DEBUG ("initializing 24-bit endian conversion");
convert->swap_endian = converter_swap_endian_24;
break;
case 32:
GST_DEBUG ("initializing 32-bit endian conversion");
convert->swap_endian = converter_swap_endian_32;
break;
case 64:
GST_DEBUG ("initializing 64-bit endian conversion");
convert->swap_endian = converter_swap_endian_64;
break;
default:
GST_ERROR ("unsupported sample width for endian conversion");
g_assert_not_reached ();
}
}
}
}
setup_allocators (convert);
return convert;
/* ERRORS */
unpositioned:
{
GST_WARNING ("unpositioned channels");
g_clear_pointer (&config, gst_structure_free);
return NULL;
}
invalid_mix_matrix:
{
GST_WARNING ("Invalid mix matrix");
g_clear_pointer (&config, gst_structure_free);
return NULL;
}
}
/**
* gst_audio_converter_free:
* @convert: a #GstAudioConverter
*
* Free a previously allocated @convert instance.
*/
void
gst_audio_converter_free (GstAudioConverter * convert)
{
AudioChain *chain;
g_return_if_fail (convert != NULL);
/* walk the chain backwards and free all elements */
for (chain = convert->chain_end; chain;) {
AudioChain *prev = chain->prev;
audio_chain_free (chain);
chain = prev;
}
if (convert->quant)
gst_audio_quantize_free (convert->quant);
if (convert->mix)
gst_audio_channel_mixer_free (convert->mix);
if (convert->resampler)
gst_audio_resampler_free (convert->resampler);
gst_audio_info_init (&convert->in);
gst_audio_info_init (&convert->out);
gst_structure_free (convert->config);
g_slice_free (GstAudioConverter, convert);
}
/**
* gst_audio_converter_get_out_frames:
* @convert: a #GstAudioConverter
* @in_frames: number of input frames
*
* Calculate how many output frames can be produced when @in_frames input
* frames are given to @convert.
*
* Returns: the number of output frames
*/
gsize
gst_audio_converter_get_out_frames (GstAudioConverter * convert,
gsize in_frames)
{
if (convert->resampler)
return gst_audio_resampler_get_out_frames (convert->resampler, in_frames);
else
return in_frames;
}
/**
* gst_audio_converter_get_in_frames:
* @convert: a #GstAudioConverter
* @out_frames: number of output frames
*
* Calculate how many input frames are currently needed by @convert to produce
* @out_frames of output frames.
*
* Returns: the number of input frames
*/
gsize
gst_audio_converter_get_in_frames (GstAudioConverter * convert,
gsize out_frames)
{
if (convert->resampler)
return gst_audio_resampler_get_in_frames (convert->resampler, out_frames);
else
return out_frames;
}
/**
* gst_audio_converter_get_max_latency:
* @convert: a #GstAudioConverter
*
* Get the maximum number of input frames that the converter would
* need before producing output.
*
* Returns: the latency of @convert as expressed in the number of
* frames.
*/
gsize
gst_audio_converter_get_max_latency (GstAudioConverter * convert)
{
if (convert->resampler)
return gst_audio_resampler_get_max_latency (convert->resampler);
else
return 0;
}
/**
* gst_audio_converter_reset:
* @convert: a #GstAudioConverter
*
* Reset @convert to the state it was when it was first created, clearing
* any history it might currently have.
*/
void
gst_audio_converter_reset (GstAudioConverter * convert)
{
if (convert->resampler)
gst_audio_resampler_reset (convert->resampler);
if (convert->quant)
gst_audio_quantize_reset (convert->quant);
}
/**
* gst_audio_converter_samples:
* @convert: a #GstAudioConverter
* @flags: extra #GstAudioConverterFlags
* @in: input frames
* @in_frames: number of input frames
* @out: output frames
* @out_frames: number of output frames
*
* Perform the conversion with @in_frames in @in to @out_frames in @out
* using @convert.
*
* In case the samples are interleaved, @in and @out must point to an
* array with a single element pointing to a block of interleaved samples.
*
* If non-interleaved samples are used, @in and @out must point to an
* array with pointers to memory blocks, one for each channel.
*
* @in may be %NULL, in which case @in_frames of silence samples are processed
* by the converter.
*
* This function always produces @out_frames of output and consumes @in_frames of
* input. Use gst_audio_converter_get_out_frames() and
* gst_audio_converter_get_in_frames() to make sure @in_frames and @out_frames
* are matching and @in and @out point to enough memory.
*
* Returns: %TRUE is the conversion could be performed.
*/
gboolean
gst_audio_converter_samples (GstAudioConverter * convert,
GstAudioConverterFlags flags, gpointer in[], gsize in_frames,
gpointer out[], gsize out_frames)
{
g_return_val_if_fail (convert != NULL, FALSE);
g_return_val_if_fail (out != NULL, FALSE);
if (in_frames == 0) {
GST_LOG ("skipping empty buffer");
return TRUE;
}
return convert->convert (convert, flags, in, in_frames, out, out_frames);
}
/**
* gst_audio_converter_convert:
* @convert: a #GstAudioConverter
* @flags: extra #GstAudioConverterFlags
* @in: (array length=in_size) (element-type guint8): input data
* @in_size: size of @in
* @out: (out) (array length=out_size) (element-type guint8): a pointer where
* the output data will be written
* @out_size: (out): a pointer where the size of @out will be written
*
* Convenience wrapper around gst_audio_converter_samples(), which will
* perform allocation of the output buffer based on the result from
* gst_audio_converter_get_out_frames().
*
* Returns: %TRUE is the conversion could be performed.
*
* Since: 1.14
*/
gboolean
gst_audio_converter_convert (GstAudioConverter * convert,
GstAudioConverterFlags flags, gpointer in, gsize in_size,
gpointer * out, gsize * out_size)
{
gsize in_frames;
gsize out_frames;
g_return_val_if_fail (convert != NULL, FALSE);
g_return_val_if_fail (flags ^ GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE, FALSE);
in_frames = in_size / convert->in.bpf;
out_frames = gst_audio_converter_get_out_frames (convert, in_frames);
*out_size = out_frames * convert->out.bpf;
*out = g_malloc0 (*out_size);
return gst_audio_converter_samples (convert, flags, &in, in_frames, out,
out_frames);
}
/**
* gst_audio_converter_supports_inplace:
* @convert: a #GstAudioConverter
*
* Returns whether the audio converter can perform the conversion in-place.
* The return value would be typically input to gst_base_transform_set_in_place()
*
* Returns: %TRUE when the conversion can be done in place.
*
* Since: 1.12
*/
gboolean
gst_audio_converter_supports_inplace (GstAudioConverter * convert)
{
return convert->in_place;
}
/**
* gst_audio_converter_is_passthrough:
*
* Returns whether the audio converter will operate in passthrough mode.
* The return value would be typically input to gst_base_transform_set_passthrough()
*
* Returns: %TRUE when no conversion will actually occur.
*
* Since: 1.16
*/
gboolean
gst_audio_converter_is_passthrough (GstAudioConverter * convert)
{
return convert->passthrough;
}
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