gst/wavparse/gstwavparse.*: reverted patch #337625 for the price of 1 hour sleep

Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
(gst_wavparse_class_init), (gst_wavparse_reset),
(gst_wavparse_init), (gst_wavparse_create_sourcepad),
(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
(gst_wavparse_send_event), (gst_wavparse_add_src_pad),
(gst_wavparse_stream_data), (gst_wavparse_loop),
(gst_wavparse_srcpad_event), (gst_wavparse_sink_activate),
(gst_wavparse_sink_activate_pull), (gst_wavparse_change_state),
(plugin_init):
* gst/wavparse/gstwavparse.h:
reverted patch #337625 for the price of 1 hour sleep

ChangeLog

@@ -1,3 +1,18 @@
+2006-04-27  Stefan Kost  <ensonic@users.sf.net>
+
+	* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
+	(gst_wavparse_class_init), (gst_wavparse_reset),
+	(gst_wavparse_init), (gst_wavparse_create_sourcepad),
+	(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
+	(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
+	(gst_wavparse_send_event), (gst_wavparse_add_src_pad),
+	(gst_wavparse_stream_data), (gst_wavparse_loop),
+	(gst_wavparse_srcpad_event), (gst_wavparse_sink_activate),
+	(gst_wavparse_sink_activate_pull), (gst_wavparse_change_state),
+	(plugin_init):
+	* gst/wavparse/gstwavparse.h:
+	  reverted patch #337625 for the price of 1 hour sleep
+
 2006-04-26  Stefan Kost  <ensonic@users.sf.net>
 
 	* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),

gst/wavparse/gstwavparse.c

@@ -1,7 +1,6 @@
 /* -*- Mode: C; tab-width: 2; indent-tabs-mode: t; c-basic-offset: 2 -*- */
 /* GStreamer
  * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
- * Copyright (C) <2006> Nokia Corporation, Stefan Kost <stefan.kost@nokia.com>.
  *
  * This library is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Library General Public
@@ -32,18 +31,11 @@
  * <title>Example launch line</title>
  * <para>
  * <programlisting>
- * gst-launch filesrc location=sine.wav ! queue ! wavparse ! audioconvert ! alsasink
+ * gst-launch filesrc sine.wav ! wavparse ! audioconvert ! alsasink
  * </programlisting>
  * Read a wav file and output to the soundcard using the ALSA element. The
  * wav file is assumed to contain raw uncompressed samples.
  * </para>
- * <para>
- * <programlisting>
- * gst-launch gnomevfssrc location=http://www.example.org/sine.wav ! wavparse ! audioconvert ! alsasink
- * </programlisting>
- * Stream data from
- * </para>
- *
  * </refsect2>
  *
  * Last reviewed on 2006-03-03 (0.10.3)
@@ -59,24 +51,18 @@
 #include "gst/riff/riff-media.h"
 #include <gst/gst-i18n-plugin.h>
 
-#ifndef G_MAXUINT32
-#define G_MAXUINT32 0xffffffff
-#endif
-
 GST_DEBUG_CATEGORY_STATIC (wavparse_debug);
 #define GST_CAT_DEFAULT (wavparse_debug)
 
 static void gst_wavparse_base_init (gpointer g_class);
 static void gst_wavparse_class_init (GstWavParseClass * klass);
 static void gst_wavparse_init (GstWavParse * wavparse);
-static void gst_wavparse_dispose (GObject * object);
 
 static gboolean gst_wavparse_sink_activate (GstPad * sinkpad);
 static gboolean gst_wavparse_sink_activate_pull (GstPad * sinkpad,
     gboolean active);
 static gboolean gst_wavparse_send_event (GstElement * element,
     GstEvent * event);
-static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstBuffer * buf);
 static GstStateChangeReturn gst_wavparse_change_state (GstElement * element,
     GstStateChange transition);
 
@@ -91,12 +77,6 @@ static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event);
 static void gst_wavparse_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec);
 
-static const GstElementDetails gst_wavparse_details =
-GST_ELEMENT_DETAILS ("WAV audio demuxer",
-    "Codec/Demuxer/Audio",
-    "Parse a .wav file into raw audio",
-    "Erik Walthinsen <omega@cse.ogi.edu>");
-
 static GstStaticPadTemplate sink_template_factory =
 GST_STATIC_PAD_TEMPLATE ("wavparse_sink",
     GST_PAD_SINK,
@@ -175,13 +155,22 @@ static void
 gst_wavparse_base_init (gpointer g_class)
 {
   GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+  GstPadTemplate *templ;
+  static GstElementDetails gst_wavparse_details =
+      GST_ELEMENT_DETAILS ("WAV audio demuxer",
+      "Codec/Demuxer/Audio",
+      "Parse a .wav file into raw audio",
+      "Erik Walthinsen <omega@cse.ogi.edu>");
+
+  gst_element_class_set_details (element_class, &gst_wavparse_details);
 
   /* register src pads */
-  gst_element_class_add_pad_template (element_class,
+  templ = gst_static_pad_template_get (&sink_template_factory);
-      gst_static_pad_template_get (&sink_template_factory));
+  gst_element_class_add_pad_template (element_class, templ);
-  gst_element_class_add_pad_template (element_class,
+  gst_object_unref (templ);
-      gst_static_pad_template_get (&src_template_factory));
+  templ = gst_static_pad_template_get (&src_template_factory);
-  gst_element_class_set_details (element_class, &gst_wavparse_details);
+  gst_element_class_add_pad_template (element_class, templ);
+  gst_object_unref (templ);
 }
 
 static void
@@ -196,28 +185,13 @@ gst_wavparse_class_init (GstWavParseClass * klass)
   parent_class = g_type_class_peek_parent (klass);
 
   object_class->get_property = gst_wavparse_get_property;
-  object_class->dispose = gst_wavparse_dispose;
 
   gstelement_class->change_state = gst_wavparse_change_state;
   gstelement_class->send_event = gst_wavparse_send_event;
+
+  GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
 }
 
-
-static void
-gst_wavparse_dispose (GObject * object)
-{
-  GST_DEBUG ("WAV: Dispose\n");
-  GstWavParse *wav = GST_WAVPARSE (object);
-
-  if (wav->adapter) {
-    g_object_unref (wav->adapter);
-    wav->adapter = NULL;
-  }
-
-  G_OBJECT_CLASS (parent_class)->dispose (object);
-}
-
-
 static void
 gst_wavparse_reset (GstWavParse * wavparse)
 {
@@ -235,12 +209,6 @@ gst_wavparse_reset (GstWavParse * wavparse)
   wavparse->dataleft = 0;
   wavparse->datasize = 0;
   wavparse->datastart = 0;
-  wavparse->got_fmt = FALSE;
-  wavparse->first = TRUE;
-
-  if (wavparse->seek_event)
-    gst_event_unref (wavparse->seek_event);
-  wavparse->seek_event = NULL;
 
   /* we keep the segment info in time */
   gst_segment_init (&wavparse->segment, GST_FORMAT_TIME);
@@ -258,12 +226,7 @@ gst_wavparse_init (GstWavParse * wavparse)
       GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate));
   gst_pad_set_activatepull_function (wavparse->sinkpad,
       GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_pull));
-  gst_pad_set_chain_function (wavparse->sinkpad,
-      GST_DEBUG_FUNCPTR (gst_wavparse_chain));
   gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->sinkpad);
-
-  /* src, will be created later */
-  wavparse->srcpad = NULL;
 }
 
 static void
@@ -291,8 +254,6 @@ gst_wavparse_create_sourcepad (GstWavParse * wavparse)
       GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));
   gst_pad_set_event_function (wavparse->srcpad,
       GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));
-
-  GST_DEBUG_OBJECT (wavparse, "srcpad created");
 }
 
 static void
@@ -309,8 +270,6 @@ gst_wavparse_get_property (GObject * object,
   }
 }
 
-
-
 #if 0
 static void
 gst_wavparse_parse_adtl (GstWavParse * wavparse, int len)
@@ -494,7 +453,9 @@ gst_wavparse_parse_adtl (GstWavParse * wavparse, int len)
 
   g_object_notify (G_OBJECT (wavparse), "metadata");
 }
+#endif
 
+#if 0
 static void
 gst_wavparse_parse_cues (GstWavParse * wavparse, int len)
 {
@@ -557,7 +518,49 @@ gst_wavparse_parse_cues (GstWavParse * wavparse, int len)
 
   g_object_notify (G_OBJECT (wavparse), "metadata");
 }
+#endif
 
+static gboolean
+gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
+{
+  guint32 doctype;
+
+  if (!gst_riff_parse_file_header (element, buf, &doctype))
+    return FALSE;
+
+  if (doctype != GST_RIFF_RIFF_WAVE)
+    goto not_wav;
+
+  return TRUE;
+
+  /* ERRORS */
+not_wav:
+  {
+    GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
+        ("File is not an WAVE file: %" GST_FOURCC_FORMAT,
+            GST_FOURCC_ARGS (doctype)));
+    return FALSE;
+  }
+}
+
+static GstFlowReturn
+gst_wavparse_stream_init (GstWavParse * wav)
+{
+  GstFlowReturn res;
+  GstBuffer *buf = NULL;
+
+  if ((res = gst_pad_pull_range (wav->sinkpad,
+              wav->offset, 12, &buf)) != GST_FLOW_OK)
+    return res;
+  else if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), buf))
+    return GST_FLOW_ERROR;
+
+  wav->offset += 12;
+
+  return GST_FLOW_OK;
+}
+
+#if 0
 /* Read 'fmt ' header */
 static gboolean
 gst_wavparse_fmt (GstWavParse * wav)
@@ -715,48 +718,6 @@ gst_wavparse_other (GstWavParse * wav)
 }
 #endif
 
-
-
-static gboolean
-gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
-{
-  guint32 doctype;
-
-  if (!gst_riff_parse_file_header (element, buf, &doctype))
-    return FALSE;
-
-  if (doctype != GST_RIFF_RIFF_WAVE)
-    goto not_wav;
-
-  return TRUE;
-
-  /* ERRORS */
-not_wav:
-  {
-    GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
-        ("File is not an WAVE file: %" GST_FOURCC_FORMAT,
-            GST_FOURCC_ARGS (doctype)));
-    return FALSE;
-  }
-}
-
-static GstFlowReturn
-gst_wavparse_stream_init (GstWavParse * wav)
-{
-  GstFlowReturn res;
-  GstBuffer *buf = NULL;
-
-  if ((res = gst_pad_pull_range (wav->sinkpad,
-              wav->offset, 12, &buf)) != GST_FLOW_OK)
-    return res;
-  else if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), buf))
-    return GST_FLOW_ERROR;
-
-  wav->offset += 12;
-
-  return GST_FLOW_OK;
-}
-
 /* This function is used to perform seeks on the element in 
  * pull mode.
  *
@@ -779,6 +740,7 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
   gboolean update;
   GstSegment seeksegment;
 
+
   if (event) {
     GST_DEBUG_OBJECT (wav, "doing seek with event");
 
@@ -808,12 +770,10 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
 
   flush = flags & GST_SEEK_FLAG_FLUSH;
 
-  if (flush && wav->srcpad) {
+  if (flush)
-    GST_DEBUG_OBJECT (wav, "sending flush start");
     gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ());
-  } else {
+  else
     gst_pad_pause_task (wav->sinkpad);
-  }
 
   GST_PAD_STREAM_LOCK (wav->sinkpad);
 
@@ -854,21 +814,18 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
       GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
 
   /* prepare for streaming again */
-  if (wav->srcpad) {
+  if (flush) {
-    if (flush) {
+    gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop ());
-      GST_DEBUG_OBJECT (wav, "sending flush stop");
+  } else if (wav->segment_running) {
-      gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop ());
+    /* we are running the current segment and doing a non-flushing seek,
-    } else if (wav->segment_running) {
+     * close the segment first based on the last_stop. */
-      /* we are running the current segment and doing a non-flushing seek,
+    GST_DEBUG_OBJECT (wav, "closing running segment %" G_GINT64_FORMAT
-       * close the segment first based on the last_stop. */
+        " to %" G_GINT64_FORMAT, wav->segment.start, wav->segment.last_stop);
-      GST_DEBUG_OBJECT (wav, "closing running segment %" G_GINT64_FORMAT
-          " to %" G_GINT64_FORMAT, wav->segment.start, wav->segment.last_stop);
 
-      gst_pad_push_event (wav->srcpad,
+    gst_pad_push_event (wav->srcpad,
-          gst_event_new_new_segment (TRUE,
+        gst_event_new_new_segment (TRUE,
-              wav->segment.rate, wav->segment.format,
+            wav->segment.rate, wav->segment.format,
-              wav->segment.start, wav->segment.last_stop, wav->segment.time));
+            wav->segment.start, wav->segment.last_stop, wav->segment.time));
-    }
   }
 
   memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
@@ -911,61 +868,6 @@ no_format:
   }
 }
 
-
-/*
- * gst_wavparse_peek_chunk_info:
- * @wav Wavparse object
- * @tag holder for tag
- * @size holder for tag size
- *                         
- * Peek next chunk info (tag and size)                         
- *
- * Returns: %TRUE when one chunk info has been got from the adapter
- */
-static gboolean
-gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)
-{
-  const guint8 *data = NULL;
-
-  if (gst_adapter_available (wav->adapter) < 8) {
-    return FALSE;
-  }
-
-  GST_DEBUG ("Next chunk size is %d bytes", *size);
-  data = gst_adapter_peek (wav->adapter, 8);
-  *tag = GST_READ_UINT32_LE (data);
-  *size = GST_READ_UINT32_LE (data + 4);
-
-  return TRUE;
-}
-
-
-/*
- * gst_wavparse_peek_chunk:
- * @wav Wavparse object
- * @tag holder for tag
- * @size holder for tag size
- *
- * Peek enough data for one full chunk
- *
- * Returns: %TRUE when one chunk has been got
- */
-static gboolean
-gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)
-{
-  guint32 peek_size = 0;
-
-  gst_wavparse_peek_chunk_info (wav, tag, size);
-  GST_DEBUG ("Need to peek chunk of %d bytes", *size);
-  peek_size = (*size + 1) & ~1;
-
-  if (gst_adapter_available (wav->adapter) >= (8 + peek_size)) {
-    return TRUE;
-  } else {
-    return FALSE;
-  }
-}
-
 static gboolean
 gst_wavparse_get_upstream_size (GstWavParse * wav, gint64 * len)
 {
@@ -985,127 +887,97 @@ static GstFlowReturn
 gst_wavparse_stream_headers (GstWavParse * wav)
 {
   GstFlowReturn res;
-  GstBuffer *buf;
+  GstBuffer *buf, *extra;
   gst_riff_strf_auds *header = NULL;
-  guint32 tag, size;
+  guint32 tag;
   gboolean gotdata = FALSE;
   GstCaps *caps;
   gint64 duration;
   gchar *codec_name = NULL;
   GstEvent **event_p;
 
-  if (!wav->got_fmt) {
+  /* The header start with a 'fmt ' tag */
-    GstBuffer *extra;
+  if ((res = gst_riff_read_chunk (GST_ELEMENT (wav), wav->sinkpad,
+              &wav->offset, &tag, &buf)) != GST_FLOW_OK)
+    return res;
 
-    /* The header start with a 'fmt ' tag */
+  else if (tag != GST_RIFF_TAG_fmt)
+    goto invalid_wav;
 
-    if (wav->streaming) {
+  if (!(gst_riff_parse_strf_auds (GST_ELEMENT (wav), buf, &header, &extra)))
-      if (!gst_wavparse_peek_chunk (wav, &tag, &size))
+    goto parse_header_error;
-        return GST_FLOW_OK;
 
-      buf = gst_buffer_new ();
+  /* Note: gst_riff_create_audio_caps might nedd to fix values in
-      gst_buffer_ref (buf);
+   * the header header depending on the format, so call it first */
-      gst_adapter_flush (wav->adapter, 8);
+  caps =
-      wav->offset += 8;
+      gst_riff_create_audio_caps (header->format, NULL, header, extra,
-      GST_BUFFER_DATA (buf) = (guint8 *) gst_adapter_peek (wav->adapter, size);
+      NULL, &codec_name);
-      GST_BUFFER_SIZE (buf) = size;
 
-    } else {
+  if (extra)
-      if ((res = gst_riff_read_chunk (GST_ELEMENT (wav), wav->sinkpad,
+    gst_buffer_unref (extra);
-                  &wav->offset, &tag, &buf)) != GST_FLOW_OK)
-        return res;
-    }
 
-    if (tag != GST_RIFF_TAG_fmt)
+  wav->format = header->format;
-      goto invalid_wav;
+  wav->rate = header->rate;
+  wav->channels = header->channels;
 
-    if (!(gst_riff_parse_strf_auds (GST_ELEMENT (wav), buf, &header, &extra)))
+  if (wav->channels == 0)
-      goto parse_header_error;
+    goto no_channels;
 
-    if (extra)
+  wav->blockalign = header->blockalign;
-      gst_buffer_unref (extra);
+  wav->width = (header->blockalign * 8) / header->channels;
+  wav->depth = header->size;
+  wav->bps = header->av_bps;
 
-    if (wav->streaming) {
+  if (wav->bps <= 0)
-      gst_adapter_flush (wav->adapter, size);
+    goto no_bitrate;
-      wav->offset += size;
-      GST_BUFFER_DATA (buf) = NULL;
-      gst_buffer_unref (buf);
-    }
 
-    /* Note: gst_riff_create_audio_caps might nedd to fix values in
+  wav->bytes_per_sample = wav->channels * wav->width / 8;
-     * the header header depending on the format, so call it first */
+  if (wav->bytes_per_sample <= 0)
-    caps =
+    goto no_bytes_per_sample;
-        gst_riff_create_audio_caps (header->format, NULL, header, NULL,
-        NULL, &codec_name);
 
-    wav->format = header->format;
+  g_free (header);
-    wav->rate = header->rate;
-    wav->channels = header->channels;
 
-    if (wav->channels == 0)
+  if (!caps)
-      goto no_channels;
+    goto unknown_format;
 
-    wav->blockalign = header->blockalign;
+  GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);
-    wav->width = (header->blockalign * 8) / header->channels;
+  GST_DEBUG_OBJECT (wav, "width      = %u", (guint) wav->width);
-    wav->depth = header->size;
+  GST_DEBUG_OBJECT (wav, "depth      = %u", (guint) wav->depth);
-    wav->bps = header->av_bps;
+  GST_DEBUG_OBJECT (wav, "bps        = %u", (guint) wav->bps);
 
-    if (wav->bps <= 0)
+  /* create pad later so we can sniff the first few bytes
-      goto no_bitrate;
+   * of the real data and correct our caps if necessary */
+  gst_caps_replace (&wav->caps, caps);
+  gst_caps_replace (&caps, NULL);
 
-    wav->bytes_per_sample = wav->channels * wav->width / 8;
+  if (codec_name) {
-    if (wav->bytes_per_sample <= 0)
+    wav->tags = gst_tag_list_new ();
-      goto no_bytes_per_sample;
 
-    g_free (header);
+    gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
+        GST_TAG_AUDIO_CODEC, codec_name, NULL);
 
-    if (!caps)
+    g_free (codec_name);
-      goto unknown_format;
+    codec_name = NULL;
-
-    GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);
-    GST_DEBUG_OBJECT (wav, "width      = %u", (guint) wav->width);
-    GST_DEBUG_OBJECT (wav, "depth      = %u", (guint) wav->depth);
-    GST_DEBUG_OBJECT (wav, "bps        = %u", (guint) wav->bps);
-
-    /* create pad later so we can sniff the first few bytes
-     * of the real data and correct our caps if necessary */
-    gst_caps_replace (&wav->caps, caps);
-    gst_caps_replace (&caps, NULL);
-
-    wav->got_fmt = TRUE;
-
-    if (codec_name) {
-      wav->tags = gst_tag_list_new ();
-
-      gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
-          GST_TAG_AUDIO_CODEC, codec_name, NULL);
-
-      g_free (codec_name);
-      codec_name = NULL;
-    }
-
-    GST_DEBUG_OBJECT (wav, "frequency %d, channels %d", wav->rate,
-        wav->channels);
   }
 
+  GST_DEBUG_OBJECT (wav, "frequency %d, channels %d", wav->rate, wav->channels);
+
   /* loop headers until we get data */
   while (!gotdata) {
-    if (wav->streaming) {
+    guint size;
-      if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
+    guint32 tag;
-        return GST_FLOW_OK;
+
-    } else {
+    if ((res =
-      if ((res =
+            gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
-              gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
+                &buf)) != GST_FLOW_OK)
-                  &buf)) != GST_FLOW_OK)
+      goto header_read_error;
-        goto header_read_error;
-      tag = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf));
-      size = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf) + 4);
-    }
 
     /*
        wav is a st00pid format, we don't know for sure where data starts.
        So we have to go bit by bit until we find the 'data' header
      */
+    tag = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf));
+    size = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf) + 4);
 
     switch (tag) {
         /* TODO : Implement the various cases */
@@ -1114,11 +986,6 @@ gst_wavparse_stream_headers (GstWavParse * wav)
 
         GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %d", size);
         gotdata = TRUE;
-        if (wav->streaming) {
-          gst_adapter_flush (wav->adapter, 8);
-        } else {
-          gst_buffer_unref (buf);
-        }
         wav->offset += 8;
         wav->datastart = wav->offset;
         /* file might be truncated */
@@ -1131,19 +998,12 @@ gst_wavparse_stream_headers (GstWavParse * wav)
         break;
       }
       default:
-        if (wav->streaming) {
-          if (!gst_wavparse_peek_chunk (wav, &tag, &size))
-            return GST_FLOW_OK;
-        }
         GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT,
             GST_FOURCC_ARGS (tag));
         wav->offset += 8 + ((size + 1) & ~1);
-        if (wav->streaming) {
+        break;
-          gst_adapter_flush (wav->adapter, 8 + ((size + 1) & ~1));
-        } else {
-          gst_buffer_unref (buf);
-        }
     }
+    gst_buffer_unref (buf);
   }
 
   GST_DEBUG_OBJECT (wav, "Finished parsing headers");
@@ -1161,7 +1021,6 @@ gst_wavparse_stream_headers (GstWavParse * wav)
   event_p = &wav->seek_event;
   gst_event_replace (event_p, NULL);
 
-  wav->state = GST_WAVPARSE_DATA;
   return GST_FLOW_OK;
 
   /* ERROR */
@@ -1221,32 +1080,6 @@ header_read_error:
   }
 }
 
-
-/*                       
- * Read WAV file tag when streaming
- */
-static GstFlowReturn
-gst_wavparse_parse_stream_init (GstWavParse * wav)
-{
-  if (gst_adapter_available (wav->adapter) >= 12) {
-    GstBuffer *tmp = gst_buffer_new ();
-
-    /* _take flushes the data */
-    GST_BUFFER_DATA (tmp) = gst_adapter_take (wav->adapter, 12);
-    GST_BUFFER_SIZE (tmp) = 12;
-
-    GST_DEBUG ("Parsing wav header");
-    if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), tmp)) {
-      return GST_FLOW_ERROR;
-    }
-
-    wav->offset += 12;
-    /* Go to next state */
-    wav->state = GST_WAVPARSE_HEADER;
-  }
-  return GST_FLOW_OK;
-}
-
 /* handle an event sent directly to the element.
  *
  * This event can be sent either in the READY state or the
@@ -1267,8 +1100,6 @@ gst_wavparse_send_event (GstElement * element, GstEvent * event)
   gboolean res = FALSE;
   GstEvent **event_p;
 
-  GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event));
-
   switch (GST_EVENT_TYPE (event)) {
     case GST_EVENT_SEEK:
       if (wav->state == GST_WAVPARSE_DATA) {
@@ -1318,7 +1149,6 @@ gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
   gst_element_add_pad (GST_ELEMENT (wav), wav->srcpad);
   gst_element_no_more_pads (GST_ELEMENT (wav));
 
-  GST_DEBUG_OBJECT (wav, "Send newsegment event on newpad");
   gst_pad_push_event (wav->srcpad, wav->newsegment);
   wav->newsegment = NULL;
 
@@ -1331,7 +1161,7 @@ gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
 #define MAX_BUFFER_SIZE 4096
 
 static GstFlowReturn
-gst_wavparse_stream_data (GstWavParse * wav)
+gst_wavparse_stream_data (GstWavParse * wav, gboolean first)
 {
   GstBuffer *buf = NULL;
   GstFlowReturn res = GST_FLOW_OK;
@@ -1339,10 +1169,8 @@ gst_wavparse_stream_data (GstWavParse * wav)
   GstClockTime timestamp, next_timestamp;
   guint64 pos, nextpos;
 
-iterate_adapter:
+  GST_LOG_OBJECT (wav, "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT,
-  GST_LOG_OBJECT (wav,
+      wav->offset, wav->end_offset);
-      "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
-      G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft);
 
   /* Get the next n bytes and output them */
   if (wav->dataleft == 0 || wav->dataleft < wav->blockalign)
@@ -1359,32 +1187,18 @@ iterate_adapter:
   GST_LOG_OBJECT (wav, "Fetching %" G_GINT64_FORMAT " bytes of data "
       "from the sinkpad", desired);
 
-  if (wav->streaming) {
+  if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
-    guint avail = gst_adapter_available (wav->adapter);
+              desired, &buf)) != GST_FLOW_OK)
+    goto pull_error;
 
-    if (avail < desired) {
+  obtained = GST_BUFFER_SIZE (buf);
-      GST_LOG_OBJECT (wav, "Got only %d bytes of data from the sinkpad", avail);
-      return GST_FLOW_OK;
-    }
-
-    buf = gst_buffer_new ();
-    GST_BUFFER_DATA (buf) = gst_adapter_take (wav->adapter, desired);
-    GST_BUFFER_SIZE (buf) = desired;
-  } else {
-    if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
-                desired, &buf)) != GST_FLOW_OK)
-      goto pull_error;
-  }
 
   /* first chunk of data? create the source pad. We do this only here so
    * we can detect broken .wav files with dts disguised as raw PCM (sigh) */
-  if (G_UNLIKELY (wav->first)) {
+  if (first) {
-    wav->first = FALSE;
     gst_wavparse_add_src_pad (wav, buf);
   }
 
-  obtained = GST_BUFFER_SIZE (buf);
-
   /* our positions */
   pos = wav->offset - wav->datastart;
   nextpos = pos + obtained;
@@ -1411,29 +1225,15 @@ iterate_adapter:
       ", size:%u", GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
       GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_BUFFER_SIZE (buf));
 
-  if (gst_pad_is_linked (wav->srcpad)) {
+  if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
-    if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
+    goto push_error;
-      goto push_error;
-  } else {
-    gst_buffer_unref (buf);
-    goto not_linked;
-  }
 
   if (obtained < wav->dataleft) {
     wav->dataleft -= obtained;
-    //wav->offset += obtained;
+    wav->offset += obtained;
   } else {
     wav->dataleft = 0;
   }
-  wav->offset += obtained;
-  /* Iterate until need more data, so adapter size won't grow */
-  if (wav->streaming) {
-    GST_LOG_OBJECT (wav,
-        "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset,
-        wav->end_offset);
-    goto iterate_adapter;
-  }
-
   return res;
 
   /* ERROR */
@@ -1467,11 +1267,6 @@ push_error:
     GST_DEBUG_OBJECT (wav, "Error pushing on srcpad");
     return res;
   }
-not_linked:
-  {
-    GST_DEBUG_OBJECT (wav, "Srcpad not linked!");
-    return GST_FLOW_ERROR;
-  }
 }
 
 static void
@@ -1480,11 +1275,8 @@ gst_wavparse_loop (GstPad * pad)
   GstFlowReturn ret;
   GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
 
-  GST_LOG_OBJECT (wav, "process data");
-
   switch (wav->state) {
     case GST_WAVPARSE_START:
-      GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_START");
       if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
         goto pause;
 
@@ -1492,15 +1284,15 @@ gst_wavparse_loop (GstPad * pad)
       /* fall-through */
 
     case GST_WAVPARSE_HEADER:
-      GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_HEADER");
       if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
         goto pause;
 
       wav->state = GST_WAVPARSE_DATA;
-      /* fall-through */
+      if ((ret = gst_wavparse_stream_data (wav, TRUE)) != GST_FLOW_OK)
-
+        goto pause;
+      break;
     case GST_WAVPARSE_DATA:
-      if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
+      if ((ret = gst_wavparse_stream_data (wav, FALSE)) != GST_FLOW_OK)
         goto pause;
       break;
     default:
@@ -1523,58 +1315,6 @@ pause:
   }
 }
 
-static GstFlowReturn
-gst_wavparse_chain (GstPad * pad, GstBuffer * buf)
-{
-  GstFlowReturn ret;
-  GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
-
-  GST_LOG_OBJECT (wav, "adapter_push %" G_GINT64_FORMAT " bytes",
-      GST_BUFFER_SIZE (buf));
-
-  gst_adapter_push (wav->adapter, buf);
-
-  switch (wav->state) {
-    case GST_WAVPARSE_START:
-      GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_START");
-      if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
-        goto pause;
-
-      wav->state = GST_WAVPARSE_HEADER;
-      /* fall-through */
-
-    case GST_WAVPARSE_HEADER:
-      GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_HEADER");
-      if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
-        goto pause;
-
-      wav->state = GST_WAVPARSE_DATA;
-      /* fall-through */
-
-    case GST_WAVPARSE_DATA:
-      if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
-        goto pause;
-      break;
-    default:
-      g_assert_not_reached ();
-  }
-
-  return ret;
-
-pause:
-  GST_LOG_OBJECT (wav, "pausing task %d", ret);
-  gst_pad_pause_task (wav->sinkpad);
-  if (GST_FLOW_IS_FATAL (ret)) {
-    /* for fatal errors we post an error message */
-    GST_ELEMENT_ERROR (wav, STREAM, FAILED,
-        (_("Internal data stream error.")),
-        ("streaming stopped, reason %s", gst_flow_get_name (ret)));
-    if (wav->srcpad != NULL)
-      gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
-  }
-  return ret;
-}
-
 #if 0
 /* convert and query stuff */
 static const GstFormat *
@@ -1786,8 +1526,7 @@ gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event)
   GstWavParse *wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
   gboolean res = TRUE;
 
-  GST_DEBUG_OBJECT (wavparse, "event %d, %s", GST_EVENT_TYPE (event),
+  GST_DEBUG_OBJECT (wavparse, "event %d", GST_EVENT_TYPE (event));
-      GST_EVENT_TYPE_NAME (event));
 
   /* can only handle events when we are in the data state */
   if (wavparse->state != GST_WAVPARSE_DATA)
@@ -1812,34 +1551,20 @@ gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event)
 static gboolean
 gst_wavparse_sink_activate (GstPad * sinkpad)
 {
-  GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (sinkpad));
+  if (gst_pad_check_pull_range (sinkpad))
-
-  if (gst_pad_check_pull_range (sinkpad)) {
-    GST_DEBUG ("going to pull mode");
-    wav->streaming = FALSE;
-    wav->adapter = NULL;
-    gst_object_unref (wav);
     return gst_pad_activate_pull (sinkpad, TRUE);
-  } else {
-    GST_DEBUG ("going to push (streaming) mode");
-    wav->streaming = TRUE;
-    wav->adapter = gst_adapter_new ();
-    gst_object_unref (wav);
-    return gst_pad_activate_push (sinkpad, TRUE);
-  }
-}
 
+  /* FIXME, we can only operate in pull mode for now */
+  GST_DEBUG_OBJECT (sinkpad, "pull_range not supported on sinkpad");
+  return FALSE;
+};
 
 static gboolean
 gst_wavparse_sink_activate_pull (GstPad * sinkpad, gboolean active)
 {
   GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (sinkpad));
 
-  GST_DEBUG_OBJECT (wav, "activating pull");
-
   if (active) {
-    /* if we have a scheduler we can start the task */
-    wav->segment_running = TRUE;
     gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop, sinkpad);
   } else {
     gst_pad_stop_task (sinkpad);
@@ -1855,8 +1580,6 @@ gst_wavparse_change_state (GstElement * element, GstStateChange transition)
   GstStateChangeReturn ret;
   GstWavParse *wav = GST_WAVPARSE (element);
 
-  GST_DEBUG_OBJECT (wav, "chaning state");
-
   switch (transition) {
     case GST_STATE_CHANGE_NULL_TO_READY:
       break;
@@ -1880,11 +1603,8 @@ gst_wavparse_change_state (GstElement * element, GstStateChange transition)
       gst_wavparse_destroy_sourcepad (wav);
       gst_event_replace (event_p, NULL);
       gst_wavparse_reset (wav);
-      if (wav->adapter) {
-        gst_adapter_clear (wav->adapter);
-      }
-      break;
     }
+      break;
     case GST_STATE_CHANGE_READY_TO_NULL:
       break;
     default:
@@ -1898,8 +1618,6 @@ plugin_init (GstPlugin * plugin)
 {
   gst_riff_init ();
 
-  GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
-
   return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY,
       GST_TYPE_WAVPARSE);
 }

gst/wavparse/gstwavparse.h

@@ -1,6 +1,5 @@
 /* GStreamer
  * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
- * Copyright (C) <2006> Nokia Corporation, Stefan Kost <stefan.kost@nokia.com>.
  *
  * This library is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Library General Public
@@ -26,7 +25,6 @@
 #include <gst/gst.h>
 #include "gst/riff/riff-ids.h"
 #include "gst/riff/riff-read.h"
-#include <gst/base/gstadapter.h>
 
 G_BEGIN_DECLS
 
@@ -95,17 +93,9 @@ struct _GstWavParse {
   /* pending seek */
   GstEvent *seek_event;
 
-  /* For streaming */
-  GstAdapter *adapter;
-  gboolean got_fmt;
-  gboolean streaming;
-
   /* configured segment, start/stop expressed in time */
   GstSegment segment;
   gboolean segment_running;
-  
-  /* for late pad configuration */
-  gboolean first;
 };
 
 struct _GstWavParseClass {