diff options
| -rw-r--r-- | ChangeLog | 9 | ||||
| -rw-r--r-- | gst/mpegaudioparse/Makefile.am | 8 | ||||
| -rw-r--r-- | gst/mpegaudioparse/gstmpegaudioparse.c | 566 | ||||
| -rw-r--r-- | gst/mpegaudioparse/gstmpegaudioparse.h | 63 | ||||
| -rw-r--r-- | gst/mpegaudioparse/mpegaudioparse.vcproj | 148 |
5 files changed, 9 insertions, 785 deletions
@@ -1,3 +1,12 @@ +2007-03-13 Jan Schmidt <thaytan@mad.scientist.com> + + * gst/mpegaudioparse/Makefile.am: + * gst/mpegaudioparse/gstmpegaudioparse.c: + * gst/mpegaudioparse/gstmpegaudioparse.h: + * gst/mpegaudioparse/mpegaudioparse.vcproj: + Remove bogus 2nd copy of mp3parse - it's actually + in -ugly. + 2007-03-12 Jan Schmidt <thaytan@mad.scientist.com> * examples/app/.cvsignore: diff --git a/gst/mpegaudioparse/Makefile.am b/gst/mpegaudioparse/Makefile.am deleted file mode 100644 index 02f8a2ee..00000000 --- a/gst/mpegaudioparse/Makefile.am +++ /dev/null @@ -1,8 +0,0 @@ -plugin_LTLIBRARIES = libgstmpegaudioparse.la - -libgstmpegaudioparse_la_SOURCES = gstmpegaudioparse.c -libgstmpegaudioparse_la_CFLAGS = $(GST_CFLAGS) -libgstmpegaudioparse_la_LIBADD = -libgstmpegaudioparse_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) - -noinst_HEADERS = gstmpegaudioparse.h diff --git a/gst/mpegaudioparse/gstmpegaudioparse.c b/gst/mpegaudioparse/gstmpegaudioparse.c deleted file mode 100644 index 0e4aa25e..00000000 --- a/gst/mpegaudioparse/gstmpegaudioparse.c +++ /dev/null @@ -1,566 +0,0 @@ -/* GStreamer - * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu> - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -/*#define GST_DEBUG_ENABLED */ -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif -#include "gstmpegaudioparse.h" - - -/* elementfactory information */ -static const GstElementDetails mp3parse_details = -GST_ELEMENT_DETAILS ("MPEG-1 audio parser", - "Codec/Parser/Audio", - "Parses and frames mpeg1 audio streams (levels 1-3), provides seek", - "Erik Walthinsen <omega@cse.ogi.edu>"); - -static GstStaticPadTemplate mp3_src_template = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/mpeg, " - "mpegversion = (int) 1, " - "layer = (int) [ 1, 3 ], " - "rate = (int) [ 8000, 48000], " "channels = (int) [ 1, 2 ]") - ); - -static GstStaticPadTemplate mp3_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) 1") - ); - -/* GstMPEGAudioParse signals and args */ -enum -{ - /* FILL ME */ - LAST_SIGNAL -}; - -enum -{ - ARG_0, - ARG_SKIP, - ARG_BIT_RATE - /* FILL ME */ -}; - - -static void gst_mp3parse_class_init (GstMPEGAudioParseClass * klass); -static void gst_mp3parse_base_init (GstMPEGAudioParseClass * klass); -static void gst_mp3parse_init (GstMPEGAudioParse * mp3parse); - -static GstFlowReturn gst_mp3parse_chain (GstPad * pad, GstBuffer * buffer); - -static int head_check (unsigned long head); - -static void gst_mp3parse_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_mp3parse_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); -static GstStateChangeReturn gst_mp3parse_change_state (GstElement * element, - GstStateChange transition); - -static GstElementClass *parent_class = NULL; - -/*static guint gst_mp3parse_signals[LAST_SIGNAL] = { 0 }; */ - -GType -gst_mp3parse_get_type (void) -{ - static GType mp3parse_type = 0; - - if (!mp3parse_type) { - static const GTypeInfo mp3parse_info = { - sizeof (GstMPEGAudioParseClass), - (GBaseInitFunc) gst_mp3parse_base_init, - NULL, - (GClassInitFunc) gst_mp3parse_class_init, - NULL, - NULL, - sizeof (GstMPEGAudioParse), - 0, - (GInstanceInitFunc) gst_mp3parse_init, - }; - - mp3parse_type = g_type_register_static (GST_TYPE_ELEMENT, - "GstMPEGAudioParse", &mp3parse_info, 0); - } - return mp3parse_type; -} - -static guint mp3types_bitrates[2][3][16] = - { {{0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,}, - {0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,}, - {0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,}}, -{{0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,}, - {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}, - {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}}, -}; - -static guint mp3types_freqs[3][3] = { {44100, 48000, 32000}, -{22050, 24000, 16000}, -{11025, 12000, 8000} -}; - -static inline guint -mp3_type_frame_length_from_header (guint32 header, guint * put_layer, - guint * put_channels, guint * put_bitrate, guint * put_samplerate) -{ - guint length; - gulong mode, samplerate, bitrate, layer, channels, padding; - gint lsf, mpg25; - - if (header & (1 << 20)) { - lsf = (header & (1 << 19)) ? 0 : 1; - mpg25 = 0; - } else { - lsf = 1; - mpg25 = 1; - } - - mode = (header >> 6) & 0x3; - channels = (mode == 3) ? 1 : 2; - samplerate = (header >> 10) & 0x3; - samplerate = mp3types_freqs[lsf + mpg25][samplerate]; - layer = 4 - ((header >> 17) & 0x3); - bitrate = (header >> 12) & 0xF; - bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000; - if (bitrate == 0) - return 0; - padding = (header >> 9) & 0x1; - switch (layer) { - case 1: - length = (bitrate * 12) / samplerate + 4 * padding; - break; - case 2: - length = (bitrate * 144) / samplerate + padding; - break; - default: - case 3: - length = (bitrate * 144) / (samplerate << lsf) + padding; - break; - } - - GST_DEBUG ("Calculated mp3 frame length of %u bytes", length); - GST_DEBUG ("samplerate = %lu, bitrate = %lu, layer = %lu, channels = %lu", - samplerate, bitrate, layer, channels); - - if (put_layer) - *put_layer = layer; - if (put_channels) - *put_channels = channels; - if (put_bitrate) - *put_bitrate = bitrate; - if (put_samplerate) - *put_samplerate = samplerate; - - return length; -} - -/* - * The chance that random data is identified as a valid mp3 header is 63 / 2^18 - * (0.024%) per try. This makes the function for calculating false positives - * 1 - (1 - ((63 / 2 ^18) ^ GST_MP3_TYPEFIND_MIN_HEADERS)) ^ buffersize) - * This has the following probabilities of false positives: - * bufsize MIN_HEADERS - * (bytes) 1 2 3 4 - * 4096 62.6% 0.02% 0% 0% - * 16384 98% 0.09% 0% 0% - * 1 MiB 100% 5.88% 0% 0% - * 1 GiB 100% 100% 1.44% 0% - * 1 TiB 100% 100% 100% 0.35% - * This means that the current choice (3 headers by most of the time 4096 byte - * buffers is pretty safe for now. - * - * The max. size of each frame is 1440 bytes, which means that for N frames - * to be detected, we need 1440 * GST_MP3_TYPEFIND_MIN_HEADERS + 3 of data. - * Assuming we step into the stream right after the frame header, this - * means we need 1440 * (GST_MP3_TYPEFIND_MIN_HEADERS + 1) - 1 + 3 bytes - * of data (5762) to always detect any mp3. - */ - -/* increase this value when this function finds too many false positives */ -#define GST_MP3_TYPEFIND_MIN_HEADERS 3 -#define GST_MP3_TYPEFIND_MIN_DATA (1440 * (GST_MP3_TYPEFIND_MIN_HEADERS + 1) - 1 + 3) - -static GstCaps * -mp3_caps_create (guint layer, guint channels, guint bitrate, guint samplerate) -{ - GstCaps *new; - - g_assert (layer); - g_assert (samplerate); - g_assert (bitrate); - g_assert (channels); - - new = gst_caps_new_simple ("audio/mpeg", - "mpegversion", G_TYPE_INT, 1, - "layer", G_TYPE_INT, layer, - "rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL); - - return new; -} - -static void -gst_mp3parse_base_init (GstMPEGAudioParseClass * klass) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (klass); - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&mp3_sink_template)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&mp3_src_template)); - gst_element_class_set_details (element_class, &mp3parse_details); -} - -static void -gst_mp3parse_class_init (GstMPEGAudioParseClass * klass) -{ - GObjectClass *gobject_class; - GstElementClass *gstelement_class; - - gobject_class = (GObjectClass *) klass; - gstelement_class = (GstElementClass *) klass; - - parent_class = g_type_class_peek_parent (klass); - - gobject_class->set_property = gst_mp3parse_set_property; - gobject_class->get_property = gst_mp3parse_get_property; - - g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP, - g_param_spec_int ("skip", "skip", "skip", - G_MININT, G_MAXINT, 0, G_PARAM_READWRITE)); - g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BIT_RATE, - g_param_spec_int ("bitrate", "Bitrate", "Bit Rate", - G_MININT, G_MAXINT, 0, G_PARAM_READABLE)); - - gstelement_class->change_state = gst_mp3parse_change_state; -} - -static void -gst_mp3parse_init (GstMPEGAudioParse * mp3parse) -{ - mp3parse->sinkpad = - gst_pad_new_from_template (gst_static_pad_template_get - (&mp3_sink_template), "sink"); - gst_pad_set_chain_function (mp3parse->sinkpad, gst_mp3parse_chain); - gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad); - - mp3parse->srcpad = - gst_pad_new_from_template (gst_static_pad_template_get - (&mp3_src_template), "src"); - gst_pad_use_fixed_caps (mp3parse->srcpad); - gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->srcpad); - /*gst_pad_set_type_id(mp3parse->srcpad, mp3frametype); */ - - mp3parse->partialbuf = NULL; - mp3parse->skip = 0; - mp3parse->in_flush = FALSE; - - mp3parse->rate = mp3parse->channels = mp3parse->layer = -1; -} - -/* FIXME, use adapter */ -static GstFlowReturn -gst_mp3parse_chain (GstPad * pad, GstBuffer * buf) -{ - GstMPEGAudioParse *mp3parse; - guchar *data; - glong size, offset = 0; - guint32 header; - int bpf; - GstBuffer *outbuf; - guint64 last_ts; - - mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad)); - - GST_DEBUG ("mp3parse: received buffer of %d bytes", GST_BUFFER_SIZE (buf)); - - last_ts = GST_BUFFER_TIMESTAMP (buf); - - /* if we have something left from the previous frame */ - if (mp3parse->partialbuf) { - GstBuffer *newbuf; - - newbuf = gst_buffer_merge (mp3parse->partialbuf, buf); - /* and the one we received.. */ - gst_buffer_unref (buf); - gst_buffer_unref (mp3parse->partialbuf); - mp3parse->partialbuf = newbuf; - } else { - mp3parse->partialbuf = buf; - } - - size = GST_BUFFER_SIZE (mp3parse->partialbuf); - data = GST_BUFFER_DATA (mp3parse->partialbuf); - - /* while we still have bytes left -4 for the header */ - while (offset < size - 4) { - int skipped = 0; - - GST_DEBUG ("mp3parse: offset %ld, size %ld ", offset, size); - - /* search for a possible start byte */ - for (; ((offset < size - 4) && (data[offset] != 0xff)); offset++) - skipped++; - if (skipped) { - GST_DEBUG ("mp3parse: **** now at %ld skipped %d bytes", offset, skipped); - } - /* construct the header word */ - header = GST_READ_UINT32_BE (data + offset); - /* if it's a valid header, go ahead and send off the frame */ - if (head_check (header)) { - guint bitrate = 0, layer = 0, rate = 0, channels = 0; - - if (!(bpf = mp3_type_frame_length_from_header (header, &layer, - &channels, &bitrate, &rate))) { - g_error ("Header failed internal error"); - } - - /******************************************************************************** - * robust seek support - * - This performs additional frame validation if the in_flush flag is set - * (indicating a discontinuous stream). - * - The current frame header is not accepted as valid unless the NEXT frame - * header has the same values for most fields. This significantly increases - * the probability that we aren't processing random data. - * - It is not clear if this is sufficient for robust seeking of Layer III - * streams which utilize the concept of a "bit reservoir" by borrow bitrate - * from previous frames. In this case, seeking may be more complicated because - * the frames are not independently coded. - ********************************************************************************/ - if (mp3parse->in_flush) { - guint32 header2; - - if ((size - offset) < (bpf + 4)) { - if (mp3parse->in_flush) - break; - } - /* wait until we have the the entire current frame as well as the next frame header */ - header2 = GST_READ_UINT32_BE (data + offset + bpf); - GST_DEBUG ("mp3parse: header=%08X, header2=%08X, bpf=%d", - (unsigned int) header, (unsigned int) header2, bpf); - -/* mask the bits which are allowed to differ between frames */ -#define HDRMASK ~((0xF << 12) /* bitrate */ | \ - (0x1 << 9) /* padding */ | \ - (0x3 << 4)) /*mode extension */ - - if ((header2 & HDRMASK) != (header & HDRMASK)) { /* require 2 matching headers in a row */ - GST_DEBUG - ("mp3parse: next header doesn't match (header=%08X, header2=%08X, bpf=%d)", - (unsigned int) header, (unsigned int) header2, bpf); - offset++; /* This frame is invalid. Start looking for a valid frame at the next position in the stream */ - continue; - } - - } - - /* if we don't have the whole frame... */ - if ((size - offset) < bpf) { - GST_DEBUG ("mp3parse: partial buffer needed %ld < %d ", (size - offset), - bpf); - break; - } else { - if (channels != mp3parse->channels || - rate != mp3parse->rate || - layer != mp3parse->layer || bitrate != mp3parse->bit_rate) { - GstCaps *caps = mp3_caps_create (layer, channels, bitrate, rate); - - gst_pad_set_caps (mp3parse->srcpad, caps); - gst_caps_unref (caps); - - mp3parse->channels = channels; - mp3parse->layer = layer; - mp3parse->rate = rate; - mp3parse->bit_rate = bitrate; - } - - outbuf = gst_buffer_create_sub (mp3parse->partialbuf, offset, bpf); - - offset += bpf; - if (mp3parse->skip == 0) { - GST_DEBUG ("mp3parse: pushing buffer of %d bytes", - GST_BUFFER_SIZE (outbuf)); - GST_BUFFER_TIMESTAMP (outbuf) = last_ts; - - if (mp3parse->layer == 1) { - GST_BUFFER_DURATION (outbuf) = 384 * GST_SECOND / mp3parse->rate; - } else { - GST_BUFFER_DURATION (outbuf) = 1152 * GST_SECOND / mp3parse->rate; - } - - gst_buffer_set_caps (outbuf, GST_PAD_CAPS (pad)); - - gst_pad_push (mp3parse->srcpad, outbuf); - - } else { - GST_DEBUG ("mp3parse: skipping buffer of %d bytes", - GST_BUFFER_SIZE (outbuf)); - gst_buffer_unref (outbuf); - mp3parse->skip--; - } - } - } else { - offset++; - GST_DEBUG ("mp3parse: *** wrong header, skipping byte (FIXME?)"); - } - } - /* if we have processed this block and there are still */ - /* bytes left not in a partial block, copy them over. */ - if (size - offset > 0) { - glong remainder = (size - offset); - - GST_DEBUG ("mp3parse: partial buffer needed %ld for trailing bytes", - remainder); - - outbuf = gst_buffer_create_sub (mp3parse->partialbuf, offset, remainder); - gst_buffer_unref (mp3parse->partialbuf); - mp3parse->partialbuf = outbuf; - } else { - gst_buffer_unref (mp3parse->partialbuf); - mp3parse->partialbuf = NULL; - } - - gst_object_unref (mp3parse); - - return GST_FLOW_OK; -} - -static gboolean -head_check (unsigned long head) -{ - GST_DEBUG ("checking mp3 header 0x%08lx", head); - /* if it's not a valid sync */ - if ((head & 0xffe00000) != 0xffe00000) { - GST_DEBUG ("invalid sync"); - return FALSE; - } - /* if it's an invalid MPEG version */ - if (((head >> 19) & 3) == 0x1) { - GST_DEBUG ("invalid MPEG version"); - return FALSE; - } - /* if it's an invalid layer */ - if (!((head >> 17) & 3)) { - GST_DEBUG ("invalid layer"); - return FALSE; - } - /* if it's an invalid bitrate */ - if (((head >> 12) & 0xf) == 0x0) { - GST_DEBUG ("invalid bitrate"); - return FALSE; - } - if (((head >> 12) & 0xf) == 0xf) { - GST_DEBUG ("invalid bitrate"); - return FALSE; - } - /* if it's an invalid samplerate */ - if (((head >> 10) & 0x3) == 0x3) { - GST_DEBUG ("invalid samplerate"); - return FALSE; - } - if ((head & 0xffff0000) == 0xfffe0000) { - GST_DEBUG ("invalid sync"); - return FALSE; - } - if (head & 0x00000002) { - GST_DEBUG ("invalid emphasis"); - return FALSE; - } - - return TRUE; -} - -static void -gst_mp3parse_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstMPEGAudioParse *src; - - g_return_if_fail (GST_IS_MP3PARSE (object)); - src = GST_MP3PARSE (object); - - switch (prop_id) { - case ARG_SKIP: - src->skip = g_value_get_int (value); - break; - default: - break; - } -} - -static void -gst_mp3parse_get_property (GObject * object, guint prop_id, GValue * value, - GParamSpec * pspec) -{ - GstMPEGAudioParse *src; - - g_return_if_fail (GST_IS_MP3PARSE (object)); - src = GST_MP3PARSE (object); - - switch (prop_id) { - case ARG_SKIP: - g_value_set_int (value, src->skip); - break; - case ARG_BIT_RATE: - g_value_set_int (value, src->bit_rate * 1000); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static GstStateChangeReturn -gst_mp3parse_change_state (GstElement * element, GstStateChange transition) -{ - GstMPEGAudioParse *src; - GstStateChangeReturn result; - - src = GST_MP3PARSE (element); - - switch (transition) { - case GST_STATE_CHANGE_PAUSED_TO_READY: - src->channels = -1; - src->rate = -1; - src->layer = -1; - break; - default: - break; - } - - result = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - - return result; -} - -static gboolean -plugin_init (GstPlugin * plugin) -{ - return gst_element_register (plugin, "mp3parse", - GST_RANK_NONE, GST_TYPE_MP3PARSE); -} - -GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, - GST_VERSION_MINOR, - "mpegaudioparse", - "MPEG-1 layer 1/2/3 audio parser", - plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN) diff --git a/gst/mpegaudioparse/gstmpegaudioparse.h b/gst/mpegaudioparse/gstmpegaudioparse.h deleted file mode 100644 index c3763c2c..00000000 --- a/gst/mpegaudioparse/gstmpegaudioparse.h +++ /dev/null @@ -1,63 +0,0 @@ -/* GStreamer - * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu> - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - - -#ifndef __MP3PARSE_H__ -#define __MP3PARSE_H__ - - -#include <gst/gst.h> - -G_BEGIN_DECLS - -#define GST_TYPE_MP3PARSE \ - (gst_mp3parse_get_type()) -#define GST_MP3PARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_MP3PARSE,GstMPEGAudioParse)) -#define GST_MP3PARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_MP3PARSE,GstMPEGAudioParseClass)) -#define GST_IS_MP3PARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_MP3PARSE)) -#define GST_IS_MP3PARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_MP3PARSE)) - -typedef struct _GstMPEGAudioParse GstMPEGAudioParse; -typedef struct _GstMPEGAudioParseClass GstMPEGAudioParseClass; - -struct _GstMPEGAudioParse { - GstElement element; - - GstPad *sinkpad,*srcpad; - - GstBuffer *partialbuf; /* previous buffer (if carryover) */ - guint skip; /* number of frames to skip */ - guint bit_rate; - gint channels, rate, layer; - gboolean in_flush; -}; - -struct _GstMPEGAudioParseClass { - GstElementClass parent_class; -}; - -GType gst_mp3parse_get_type(void); - -G_END_DECLS - -#endif /* __MP3PARSE_H__ */ diff --git a/gst/mpegaudioparse/mpegaudioparse.vcproj b/gst/mpegaudioparse/mpegaudioparse.vcproj deleted file mode 100644 index 198e78e4..00000000 --- a/gst/mpegaudioparse/mpegaudioparse.vcproj +++ /dev/null @@ -1,148 +0,0 @@ -<?xml version="1.0" encoding="Windows-1252"?> -<VisualStudioProject - ProjectType="Visual C++" - Version="7.10" - Name="mpegaudioparse" - ProjectGUID="{979C216F-0ACF-4956-AE00-055A42D678BD}" - Keyword="Win32Proj"> - <Platforms> - <Platform - Name="Win32"/> - </Platforms> - <Configurations> - <Configuration - Name="Debug|Win32" - OutputDirectory="../../win32/Debug" - IntermediateDirectory="../../win32/Debug" - 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