Move the audio-converter helper to the audio library.
$(top_srcdir)/ext/vorbis/gstvorbisparse.h \
$(top_srcdir)/ext/vorbis/gstvorbistag.h \
$(top_srcdir)/gst/adder/gstadder.h \
- $(top_srcdir)/gst/audioconvert/audioconvert.h \
$(top_srcdir)/gst/audioconvert/gstaudioconvert.h \
$(top_srcdir)/gst/audiotestsrc/gstaudiotestsrc.h \
$(top_srcdir)/gst/encoding/gstencodebin.h \
audio-format.h \
audio-channels.h \
audio-channel-mix.h \
+ audio-converter.h \
audio-info.h \
audio-quantize.h \
gstaudioringbuffer.h
audio-format.c \
audio-channels.c \
audio-channel-mix.c \
+ audio-converter.c \
audio-info.c \
audio-quantize.c \
gstaudioringbuffer.c \
audio-format.h \
audio-channels.h \
audio-channel-mix.h \
+ audio-converter.h \
audio-info.h \
audio-quantize.h \
gstaudioringbuffer.h \
--- /dev/null
+/* 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);
+
+ 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
+ * @samples: number of 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, gint 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 (samples >= 0, FALSE);
+
+ if (samples == 0)
+ return TRUE;
+
+ if (convert->passthrough) {
+ memcpy (dst, src, samples * convert->in.bpf);
+ return TRUE;
+ }
+
+ size =
+ sizeof (gdouble) * 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,
+ 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, 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, 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, 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, samples);
+ src = outbuf;
+ }
+
+ /* step 6, pack */
+ if (!convert->out_default) {
+ convert->out.finfo->pack_func (convert->out.finfo, 0, src, dst,
+ samples * convert->out.channels);
+ }
+
+ return TRUE;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * (C) 2015 Wim Taymans <wim.taymans@gmail.com>
+ *
+ * audioconverter.h: audio format conversion library
+ *
+ * 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.
+ */
+
+#ifndef __GST_AUDIO_CONVERTER_H__
+#define __GST_AUDIO_CONVERTER_H__
+
+#include <gst/gst.h>
+#include <gst/audio/audio.h>
+
+typedef struct _GstAudioConverter GstAudioConverter;
+
+/**
+ * GST_AUDIO_CONVERTER_OPT_DITHER_METHOD:
+ *
+ * #GST_TYPE_AUDIO_DITHER_METHOD, The dither method to use when
+ * changing bit depth.
+ * Default is #GST_AUDIO_DITHER_NONE.
+ */
+#define GST_AUDIO_CONVERTER_OPT_DITHER_METHOD "GstAudioConverter.dither-method"
+
+/**
+ * GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD:
+ *
+ * #GST_TYPE_AUDIO_NOISE_SHAPING_METHOD, The noise shaping method to use
+ * to mask noise from quantization errors.
+ * Default is #GST_AUDIO_NOISE_SHAPING_NONE.
+ */
+#define GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD "GstAudioConverter.noise-shaping-method"
+
+/**
+ * GST_AUDIO_CONVERTER_OPT_QUANTIZATION:
+ *
+ * #G_TYPE_UINT, The quantization amount. Components will be
+ * quantized to multiples of this value.
+ * Default is 1
+ */
+#define GST_AUDIO_CONVERTER_OPT_QUANTIZATION "GstAudioConverter.quantization"
+
+
+/**
+ * @GST_AUDIO_CONVERTER_FLAG_NONE: no flag
+ * @GST_AUDIO_CONVERTER_FLAG_SOURCE_WRITABLE: the source is writable and can be
+ * used as temporary storage during conversion.
+ *
+ * Extra flags passed to gst_audio_converter_samples().
+ */
+typedef enum {
+ GST_AUDIO_CONVERTER_FLAG_NONE = 0,
+ GST_AUDIO_CONVERTER_FLAG_SOURCE_WRITABLE = (1 << 0)
+} GstAudioConverterFlags;
+
+GstAudioConverter * gst_audio_converter_new (GstAudioInfo *in_info,
+ GstAudioInfo *out_info,
+ GstStructure *config);
+
+void gst_audio_converter_free (GstAudioConverter * convert);
+
+gboolean gst_audio_converter_set_config (GstAudioConverter * convert, GstStructure *config);
+const GstStructure * gst_audio_converter_get_config (GstAudioConverter * convert);
+
+
+gboolean gst_audio_converter_samples (GstAudioConverter * convert,
+ GstAudioConverterFlags flags,
+ gpointer src, gpointer dst,
+ gint samples);
+
+#endif /* __GST_AUDIO_CONVERTER_H__ */
#include <gst/audio/audio-channel-mix.h>
#include <gst/audio/audio-info.h>
#include <gst/audio/audio-quantize.h>
+#include <gst/audio/audio-converter.h>
G_BEGIN_DECLS
mulll t, r, 1103515245
addl r, t, 12345
+.function audio_orc_s32_to_double
+.dest 8 d1 gdouble
+.source 4 s1 gint32
+.temp 8 t1
+
+convld t1, s1
+divd d1, t1, 2147483648.0L
+
+.function audio_orc_double_to_s32
+.dest 4 d1 gint32
+.source 8 s1 gdouble
+.temp 8 t1
+
+muld t1, s1, 2147483648.0L
+convdl d1, t1
plugin_LTLIBRARIES = libgstaudioconvert.la
-ORC_SOURCE=gstaudioconvertorc
-include $(top_srcdir)/common/orc.mak
-
libgstaudioconvert_la_SOURCES = \
gstaudioconvert.c \
- audioconvert.c \
plugin.c
-nodist_libgstaudioconvert_la_SOURCES = $(ORC_NODIST_SOURCES)
-libgstaudioconvert_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(ORC_CFLAGS)
+libgstaudioconvert_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
libgstaudioconvert_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstaudioconvert_la_LIBADD = \
$(top_builddir)/gst-libs/gst/audio/libgstaudio-@GST_API_VERSION@.la \
- $(GST_BASE_LIBS) $(GST_LIBS) $(ORC_LIBS) $(LIBM)
+ $(GST_BASE_LIBS) $(GST_LIBS) $(LIBM)
libgstaudioconvert_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = \
gstaudioconvert.h \
- audioconvert.h \
plugin.h
+++ /dev/null
-/* 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 "audioconvert.h"
-#include "gstaudioconvertorc.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);
-
- 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_convert_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_convert_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
- * @samples: number of 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, gint 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 (samples >= 0, FALSE);
-
- if (samples == 0)
- return TRUE;
-
- if (convert->passthrough) {
- memcpy (dst, src, samples * convert->in.bpf);
- return TRUE;
- }
-
- size =
- sizeof (gdouble) * 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,
- 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, 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, 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, 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, samples);
- src = outbuf;
- }
-
- /* step 6, pack */
- if (!convert->out_default) {
- convert->out.finfo->pack_func (convert->out.finfo, 0, src, dst,
- samples * convert->out.channels);
- }
-
- return TRUE;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
- * (C) 2015 Wim Taymans <wim.taymans@gmail.com>
- *
- * audioconverter.h: audio format conversion library
- *
- * 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.
- */
-
-#ifndef __GST_AUDIO_CONVERTER_H__
-#define __GST_AUDIO_CONVERTER_H__
-
-#include <gst/gst.h>
-#include <gst/audio/audio.h>
-
-typedef struct _GstAudioConverter GstAudioConverter;
-
-/**
- * GST_AUDIO_CONVERTER_OPT_DITHER_METHOD:
- *
- * #GST_TYPE_AUDIO_DITHER_METHOD, The dither method to use when
- * changing bit depth.
- * Default is #GST_AUDIO_DITHER_NONE.
- */
-#define GST_AUDIO_CONVERTER_OPT_DITHER_METHOD "GstAudioConverter.dither-method"
-
-/**
- * GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD:
- *
- * #GST_TYPE_AUDIO_NOISE_SHAPING_METHOD, The noise shaping method to use
- * to mask noise from quantization errors.
- * Default is #GST_AUDIO_NOISE_SHAPING_NONE.
- */
-#define GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD "GstAudioConverter.noise-shaping-method"
-
-/**
- * GST_AUDIO_CONVERTER_OPT_QUANTIZATION:
- *
- * #G_TYPE_UINT, The quantization amount. Components will be
- * quantized to multiples of this value.
- * Default is 1
- */
-#define GST_AUDIO_CONVERTER_OPT_QUANTIZATION "GstAudioConverter.quantization"
-
-
-/**
- * @GST_AUDIO_CONVERTER_FLAG_NONE: no flag
- * @GST_AUDIO_CONVERTER_FLAG_SOURCE_WRITABLE: the source is writable and can be
- * used as temporary storage during conversion.
- *
- * Extra flags passed to gst_audio_converter_samples().
- */
-typedef enum {
- GST_AUDIO_CONVERTER_FLAG_NONE = 0,
- GST_AUDIO_CONVERTER_FLAG_SOURCE_WRITABLE = (1 << 0)
-} GstAudioConverterFlags;
-
-GstAudioConverter * gst_audio_converter_new (GstAudioInfo *in_info,
- GstAudioInfo *out_info,
- GstStructure *config);
-
-void gst_audio_converter_free (GstAudioConverter * convert);
-
-gboolean gst_audio_converter_set_config (GstAudioConverter * convert, GstStructure *config);
-const GstStructure * gst_audio_converter_get_config (GstAudioConverter * convert);
-
-
-gboolean gst_audio_converter_samples (GstAudioConverter * convert,
- GstAudioConverterFlags flags,
- gpointer src, gpointer dst,
- gint samples);
-
-#endif /* __GST_AUDIO_CONVERTER_H__ */
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
-#include "audioconvert.h"
-
#define GST_TYPE_AUDIO_CONVERT (gst_audio_convert_get_type())
#define GST_AUDIO_CONVERT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CONVERT,GstAudioConvert))
#define GST_AUDIO_CONVERT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_CONVERT,GstAudioConvertClass))
endif
if HAVE_ORC
-check_orc = orc/video orc/audio orc/adder orc/audioconvert orc/volume orc/videotestsrc
+check_orc = orc/video orc/audio orc/adder orc/volume orc/videotestsrc
else
check_orc =
endif
orc_adder_CFLAGS = $(ORC_CFLAGS)
orc_adder_LDADD = $(ORC_LIBS) -lorc-test-0.4
nodist_orc_adder_SOURCES = orc/adder.c
-orc_audioconvert_CFLAGS = $(ORC_CFLAGS)
-orc_audioconvert_LDADD = $(ORC_LIBS) -lorc-test-0.4
-nodist_orc_audioconvert_SOURCES = orc/audioconvert.c
orc_volume_CFLAGS = $(ORC_CFLAGS)
orc_volume_LDADD = $(ORC_LIBS) -lorc-test-0.4
nodist_orc_volume_SOURCES = orc/volume.c
$(MKDIR_P) orc/
$(ORCC) --test -o $@ $<
-orc/audioconvert.c: $(top_srcdir)/gst/audioconvert/gstaudioconvertorc.orc
- $(MKDIR_P) orc/
- $(ORCC) --test -o $@ $<
-
orc/volume.c: $(top_srcdir)/gst/volume/gstvolumeorc.orc
$(MKDIR_P) orc/
$(ORCC) --test -o $@ $<
gst_audio_clock_invalidate
gst_audio_clock_new
gst_audio_clock_reset
+ gst_audio_converter_flags_get_type
+ gst_audio_converter_free
+ gst_audio_converter_get_config
+ gst_audio_converter_new
+ gst_audio_converter_samples
+ gst_audio_converter_set_config
gst_audio_decoder_allocate_output_buffer
gst_audio_decoder_finish_frame
gst_audio_decoder_get_allocator
projects / platform / upstream / gstreamer.git / commitdiff