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gst/audioconvert/: Support for 64-bit float audio in audioconvert (#339837)
Original commit message from CVS: * gst/audioconvert/audioconvert.c: (float), (double), (audio_convert_get_func_index): * gst/audioconvert/gstaudioconvert.c: (set_structure_widths), (make_lossless_changes): Support for 64-bit float audio in audioconvert (#339837)
This commit is contained in:
Stefan Kost
parent
6f25185e75
commit
36180a5f3d
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@ -1,3 +1,11 @@
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2007-02-02 Stefan Kost <ensonic@users.sf.net>
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* gst/audioconvert/audioconvert.c: (float), (double),
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(audio_convert_get_func_index):
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* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
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(make_lossless_changes):
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Support for 64-bit float audio in audioconvert (#339837)
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2007-02-01 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Holger Wansing <linux wansing-online de>
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@ -28,8 +28,9 @@
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#include "gstchannelmix.h"
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#include "audioconvert.h"
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/* int to float conversion: int2float(i) = 1 / (2^31-1) * i */
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/* int to float/double conversion: int2xxx(i) = 1 / (2^31-1) * i */
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#define INT2FLOAT(i) (4.6566128752457969e-10 * ((gfloat)i))
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#define INT2DOUBLE(i) (4.6566128752457969e-10 * ((gdouble)i))
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/* sign bit in the intermediate format */
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#define SIGNED (1U<<31)
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@ -52,7 +53,7 @@ MAKE_UNPACK_FUNC_NAME (name) (gpointer src, gint32 *dst, \
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} \
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}
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/* special unpack code for float */
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/* special unpack code for float/double */
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static void
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MAKE_UNPACK_FUNC_NAME (float) (gpointer src, gint32 * dst,
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gint scale, gint count)
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@ -61,6 +62,21 @@ MAKE_UNPACK_FUNC_NAME (float) (gpointer src, gint32 * dst,
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gdouble temp;
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for (; count; count--) {
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/* blow up to 32 bit */
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temp = (*p++ * 2147483647.0) + 0.5;
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*dst++ = (gint32) CLAMP (temp, G_MININT32, G_MAXINT32);
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}
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}
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static void
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MAKE_UNPACK_FUNC_NAME (double) (gpointer src, gint32 * dst,
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gint scale, gint count)
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{
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gdouble *p = (gdouble *) src;
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gdouble temp;
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for (; count; count--) {
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/* blow up to 32 bit */
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temp = (*p++ * 2147483647.0) + 0.5;
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*dst++ = (gint32) CLAMP (temp, G_MININT32, G_MAXINT32);
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}
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@ -109,7 +125,7 @@ MAKE_PACK_FUNC_NAME (name) (gint32 *src, gpointer dst, \
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} \
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}
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/* special float pack function */
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/* special pack code for float/double */
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static void
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MAKE_PACK_FUNC_NAME (float) (gint32 * src, gpointer dst, gint scale, gint count)
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{
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@ -120,6 +136,17 @@ MAKE_PACK_FUNC_NAME (float) (gint32 * src, gpointer dst, gint scale, gint count)
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}
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}
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static void
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MAKE_PACK_FUNC_NAME (double) (gint32 * src, gpointer dst, gint scale,
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gint count)
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{
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gdouble *p = (gdouble *) dst;
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for (; count; count--) {
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*p++ = INT2DOUBLE (*src++);
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}
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}
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#define WRITE8(p, v) GST_WRITE_UINT8 (p, v)
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#define WRITE16_TO_LE(p,v) GST_WRITE_UINT16_LE (p, (guint16)(v))
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#define WRITE16_TO_BE(p,v) GST_WRITE_UINT16_BE (p, (guint16)(v))
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@ -161,6 +188,7 @@ static AudioConvertUnpack unpack_funcs[] = {
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MAKE_UNPACK_FUNC_NAME (u32_be),
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MAKE_UNPACK_FUNC_NAME (s32_be),
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MAKE_UNPACK_FUNC_NAME (float),
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MAKE_UNPACK_FUNC_NAME (double),
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};
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static AudioConvertPack pack_funcs[] = {
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@ -181,6 +209,7 @@ static AudioConvertPack pack_funcs[] = {
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MAKE_PACK_FUNC_NAME (u32_be),
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MAKE_PACK_FUNC_NAME (s32_be),
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MAKE_PACK_FUNC_NAME (float),
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MAKE_PACK_FUNC_NAME (double),
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};
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static gint
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@ -193,7 +222,8 @@ audio_convert_get_func_index (AudioConvertFmt * fmt)
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index += fmt->endianness == G_LITTLE_ENDIAN ? 0 : 2;
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index += fmt->sign ? 1 : 0;
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} else {
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index = 16;
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/* this is float/double */
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index = (fmt->width == 32) ? 16 : 17;
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}
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return index;
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}
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@ -120,6 +120,11 @@ GST_BOILERPLATE_FULL (GstAudioConvert, gst_audio_convert, GstBaseTransform,
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#define STATIC_CAPS \
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GST_STATIC_CAPS ( \
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"audio/x-raw-float, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) 64;" \
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"audio/x-raw-float, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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@ -316,6 +321,30 @@ parse_error:
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}
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}
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/* Set widths (a list); multiples of 8 between min and max */
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static void
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set_structure_widths (GstStructure * s, int min, int max)
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{
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GValue list = { 0 };
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GValue val = { 0 };
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int width;
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if (min == max) {
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gst_structure_set (s, "width", G_TYPE_INT, min, NULL);
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return;
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}
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g_value_init (&list, GST_TYPE_LIST);
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g_value_init (&val, G_TYPE_INT);
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for (width = min; width <= max; width += 8) {
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g_value_set_int (&val, width);
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gst_value_list_append_value (&list, &val);
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}
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gst_structure_set_value (s, "width", &list);
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g_value_unset (&val);
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g_value_unset (&list);
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}
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/* Modify the structure so that things that must always have a single
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* value (for float), or can always be losslessly converted (for int), have
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* appropriate values.
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@ -325,10 +354,10 @@ make_lossless_changes (GstStructure * s, gboolean isfloat)
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{
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if (isfloat) {
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/* float doesn't have a depth or signedness field and only supports a
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* width of 32 and native endianness */
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* widths of 32/64 and native endianness */
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gst_structure_remove_field (s, "depth");
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gst_structure_remove_field (s, "signed");
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gst_structure_set (s, "width", G_TYPE_INT, 32, NULL);
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set_structure_widths (s, 32, 64);
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gst_structure_set (s, "endianness", G_TYPE_INT, G_BYTE_ORDER, NULL);
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} else {
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/* int supports either endian, and signed or unsigned. GValues are a pain */
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@ -381,30 +410,6 @@ append_with_other_format (GstCaps * caps, GstStructure * s, gboolean isfloat)
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}
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}
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/* Set widths (a list); multiples of 8 between min and max */
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static void
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set_structure_widths (GstStructure * s, int min, int max)
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{
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GValue list = { 0 };
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GValue val = { 0 };
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int width;
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if (min == max) {
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gst_structure_set (s, "width", G_TYPE_INT, min, NULL);
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return;
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}
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g_value_init (&list, GST_TYPE_LIST);
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g_value_init (&val, G_TYPE_INT);
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for (width = min; width <= max; width += 8) {
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g_value_set_int (&val, width);
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gst_value_list_append_value (&list, &val);
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}
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gst_structure_set_value (s, "width", &list);
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g_value_unset (&val);
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g_value_unset (&list);
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}
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/* Audioconvert can perform all conversions on audio except for resampling.
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* However, there are some conversions we _prefer_ not to do. For example, it's
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* better to convert format (float<->int, endianness, etc) than the number of
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