gst/mpegaudioparse/: Remove bogus 2nd copy of mp3parse - it's actually in -ugly.

Original commit message from CVS:
* 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.

1 files changed, 0 insertions, 566 deletions

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)