gstreamer/gst-libs/gst/audio/audio-converter.c
Luis de Bethencourt df16e8dd5a audio-converter: remove unneeded check for unsigned < 0
Commit ff6d1a2a25 changed sample's type from
gint to gsize (and renamed it to in_samples). gsize is an unsigned long,
which means it can never be a negative value and the check making sure that
in_samples is >= 0 is never going to be false. Removing it.

CID 1338689
2015-11-12 14:18:30 +00:00

490 lines
14 KiB
C

/* 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:audioconverter
* @short_description: Generic audio conversion
*
* <refsect2>
* <para>
* This object is used to convert audio samples from one format to another.
* The object can perform conversion of:
* <itemizedlist>
* <listitem><para>
* audio format with optional dithering and noise shaping
* </para></listitem>
* <listitem><para>
* audio samplerate
* </para></listitem>
* <listitem><para>
* audio channels and channel layout
* </para></listitem>
* </para>
* </refsect2>
*/
#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 void (*AudioConvertFunc) (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
*/
struct _GstAudioConverter
{
GstAudioInfo in;
GstAudioInfo out;
GstStructure *config;
gboolean in_default;
AudioConvertFunc convert_in;
gboolean mix_passthrough;
GstAudioChannelMix *mix;
AudioConvertFunc convert_out;
GstAudioQuantize *quant;
gboolean out_default;
gboolean passthrough;
gpointer tmpbuf;
gpointer tmpbuf2;
gint tmpbufsize;
};
/*
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;
}
#define DEFAULT_OPT_DITHER_METHOD GST_AUDIO_DITHER_NONE
#define DEFAULT_OPT_NOISE_SHAPING_METHOD GST_AUDIO_NOISE_SHAPING_NONE
#define DEFAULT_OPT_QUANTIZATION 1
#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_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)
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_set_config:
* @convert: a #GstAudioConverter
* @config: (transfer full): a #GstStructure
*
* Set @config as extra configuraion for @convert.
*
* 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 @config could be set.
*/
gboolean
gst_audio_converter_set_config (GstAudioConverter * convert,
GstStructure * config)
{
g_return_val_if_fail (convert != NULL, FALSE);
g_return_val_if_fail (config != NULL, FALSE);
gst_structure_foreach (config, copy_config, convert);
gst_structure_free (config);
return TRUE;
}
/**
* gst_audio_converter_get_config:
* @convert: a #GstAudioConverter
*
* Get the current configuration of @convert.
*
* Returns: a #GstStructure that remains valid for as long as @convert is valid
* or until gst_audio_converter_set_config() is called.
*/
const GstStructure *
gst_audio_converter_get_config (GstAudioConverter * convert)
{
g_return_val_if_fail (convert != NULL, NULL);
return convert->config;
}
/**
* gst_audio_converter_new: (skip)
* @in: a source #GstAudioInfo
* @out: a destination #GstAudioInfo
* @config: (transfer full): 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_VIDEO_CONVERTER_OPT_*
* parameters for details about the options and values.
*
* Returns: a #GstAudioConverter or %NULL if conversion is not possible.
*/
GstAudioConverter *
gst_audio_converter_new (GstAudioInfo * in, GstAudioInfo * out,
GstStructure * config)
{
GstAudioConverter *convert;
gint in_depth, out_depth;
GstAudioChannelMixFlags flags;
gboolean in_int, out_int;
GstAudioFormat format;
GstAudioDitherMethod dither;
GstAudioNoiseShapingMethod ns;
g_return_val_if_fail (in != NULL, FALSE);
g_return_val_if_fail (out != NULL, FALSE);
g_return_val_if_fail (in->rate == out->rate, FALSE);
g_return_val_if_fail (in->layout == GST_AUDIO_LAYOUT_INTERLEAVED, FALSE);
g_return_val_if_fail (in->layout == out->layout, FALSE);
if ((GST_AUDIO_INFO_CHANNELS (in) != GST_AUDIO_INFO_CHANNELS (out)) &&
(GST_AUDIO_INFO_IS_UNPOSITIONED (in)
|| GST_AUDIO_INFO_IS_UNPOSITIONED (out)))
goto unpositioned;
convert = g_slice_new0 (GstAudioConverter);
convert->in = *in;
convert->out = *out;
/* default config */
convert->config = gst_structure_new_empty ("GstAudioConverter");
if (config)
gst_audio_converter_set_config (convert, config);
dither = GET_OPT_DITHER_METHOD (convert);
ns = GET_OPT_NOISE_SHAPING_METHOD (convert);
GST_INFO ("unitsizes: %d -> %d", in->bpf, out->bpf);
in_depth = GST_AUDIO_FORMAT_INFO_DEPTH (in->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 (in->finfo);
out_int = GST_AUDIO_FORMAT_INFO_IS_INTEGER (out->finfo);
flags =
GST_AUDIO_INFO_IS_UNPOSITIONED (in) ?
GST_AUDIO_CHANNEL_MIX_FLAGS_UNPOSITIONED_IN : 0;
flags |=
GST_AUDIO_INFO_IS_UNPOSITIONED (out) ?
GST_AUDIO_CHANNEL_MIX_FLAGS_UNPOSITIONED_OUT : 0;
/* step 1, unpack */
format = in->finfo->unpack_format;
convert->in_default = in->finfo->unpack_format == in->finfo->format;
GST_INFO ("unpack format %s to %s",
gst_audio_format_to_string (in->finfo->format),
gst_audio_format_to_string (format));
/* step 2, optional convert from S32 to F64 for channel mix */
if (in_int && !out_int) {
GST_INFO ("convert S32 to F64");
convert->convert_in = (AudioConvertFunc) audio_orc_s32_to_double;
format = GST_AUDIO_FORMAT_F64;
}
/* step 3, channel mix */
convert->mix =
gst_audio_channel_mix_new (flags, format, in->channels, in->position,
out->channels, out->position);
convert->mix_passthrough =
gst_audio_channel_mix_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);
/* step 4, optional convert for quantize */
if (!in_int && out_int) {
GST_INFO ("convert F64 to S32");
convert->convert_out = (AudioConvertFunc) audio_orc_double_to_s32;
format = GST_AUDIO_FORMAT_S32;
}
/* step 5, optional quantize */
/* 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 > 20 || (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) {
GST_INFO ("quantize to %d bits, dither %d, ns %d", out_depth, dither, ns);
convert->quant = gst_audio_quantize_new (dither, ns, 0, format,
out->channels, 1U << (32 - out_depth));
}
/* step 6, pack */
g_assert (out->finfo->unpack_format == 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));
/* optimize */
if (out->finfo->format == in->finfo->format && convert->mix_passthrough) {
GST_INFO ("same formats and passthrough mixing -> passthrough");
convert->passthrough = TRUE;
}
return convert;
/* ERRORS */
unpositioned:
{
GST_WARNING ("unpositioned channels");
return NULL;
}
}
/**
* gst_audio_converter_free:
* @convert: a #GstAudioConverter
*
* Free a previously allocated @convert instance.
*/
void
gst_audio_converter_free (GstAudioConverter * convert)
{
g_return_if_fail (convert != NULL);
if (convert->quant)
gst_audio_quantize_free (convert->quant);
if (convert->mix)
gst_audio_channel_mix_free (convert->mix);
gst_audio_info_init (&convert->in);
gst_audio_info_init (&convert->out);
g_free (convert->tmpbuf);
g_free (convert->tmpbuf2);
gst_structure_free (convert->config);
g_slice_free (GstAudioConverter, convert);
}
/**
* gst_audio_converter_samples:
* @convert: a #GstAudioConverter
* @flags: extra #GstAudioConverterFlags
* @src: source samples
* @dst: output samples
* @in_samples: number of input samples
* @out_samples: number of output samples
*
* Perform the conversion @src to @dst using @convert.
*
* Returns: %TRUE is the conversion could be performed.
*/
gboolean
gst_audio_converter_samples (GstAudioConverter * convert,
GstAudioConverterFlags flags, gpointer src, gpointer dst,
gsize in_samples, gsize * out_samples)
{
guint size;
gpointer outbuf, tmpbuf, tmpbuf2;
g_return_val_if_fail (convert != NULL, FALSE);
g_return_val_if_fail (src != NULL, FALSE);
g_return_val_if_fail (dst != NULL, FALSE);
g_return_val_if_fail (out_samples != NULL, FALSE);
if (in_samples == 0) {
*out_samples = 0;
return TRUE;
}
if (convert->passthrough) {
memcpy (dst, src, in_samples * convert->in.bpf);
*out_samples = in_samples;
return TRUE;
}
size =
sizeof (gdouble) * in_samples * MAX (convert->in.channels,
convert->out.channels);
if (size > convert->tmpbufsize) {
convert->tmpbuf = g_realloc (convert->tmpbuf, size);
convert->tmpbuf2 = g_realloc (convert->tmpbuf2, size);
convert->tmpbufsize = size;
}
tmpbuf = convert->tmpbuf;
tmpbuf2 = convert->tmpbuf2;
/* 1. unpack */
if (!convert->in_default) {
if (!convert->convert_in && convert->mix_passthrough
&& !convert->convert_out && !convert->quant && convert->out_default)
outbuf = dst;
else
outbuf = tmpbuf;
convert->in.finfo->unpack_func (convert->in.finfo,
GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE, outbuf, src,
in_samples * convert->in.channels);
src = outbuf;
}
/* 2. optionally convert for mixing */
if (convert->convert_in) {
if (convert->mix_passthrough && !convert->convert_out && !convert->quant
&& convert->out_default)
outbuf = dst;
else if (src == tmpbuf)
outbuf = tmpbuf2;
else
outbuf = tmpbuf;
convert->convert_in (outbuf, src, in_samples * convert->in.channels);
src = outbuf;
}
/* step 3, channel mix if not passthrough */
if (!convert->mix_passthrough) {
if (!convert->convert_out && !convert->quant && convert->out_default)
outbuf = dst;
else
outbuf = tmpbuf;
gst_audio_channel_mix_samples (convert->mix, src, outbuf, in_samples);
src = outbuf;
}
/* step 4, optional convert F64 -> S32 for quantize */
if (convert->convert_out) {
if (!convert->quant && convert->out_default)
outbuf = dst;
else
outbuf = tmpbuf;
convert->convert_out (outbuf, src, in_samples * convert->out.channels);
src = outbuf;
}
/* step 5, optional quantize */
if (convert->quant) {
if (convert->out_default)
outbuf = dst;
else
outbuf = tmpbuf;
gst_audio_quantize_samples (convert->quant, outbuf, src, in_samples);
src = outbuf;
}
/* step 6, pack */
if (!convert->out_default) {
convert->out.finfo->pack_func (convert->out.finfo, 0, src, dst,
in_samples * convert->out.channels);
}
*out_samples = in_samples;
return TRUE;
}