From: Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
Date: Tue, 1 Mar 2011 10:56:29 +0000 (+0100)
Subject: Added audio directory for audio codec base classes
X-Git-Tag: 1.19.3~511^2~6555^2~595
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=fc6b42122781bb85ad839b83d54eae2fc3161961;p=platform%2Fupstream%2Fgstreamer.git

Added audio directory for audio codec base classes
---

diff --git a/gst-libs/gst/audio/gstbaseaudiodecoder.c b/gst-libs/gst/audio/gstbaseaudiodecoder.c
new file mode 100644
index 0000000..8e942bf
--- /dev/null
+++ b/gst-libs/gst/audio/gstbaseaudiodecoder.c
@@ -0,0 +1,1174 @@
+/* GStreamer
+ * Copyright (C) 2009 Igalia S.L.
+ * Author: Iago Toral <itoral@igalia.com>
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstbaseaudiodecoder.h"
+
+#include <string.h>
+
+GST_DEBUG_CATEGORY_EXTERN (baseaudio_debug);
+#define GST_CAT_DEFAULT baseaudio_debug
+
+static void gst_base_audio_decoder_finalize (GObject * object);
+
+static gboolean gst_base_audio_decoder_sink_setcaps (GstPad * pad,
+    GstCaps * caps);
+static gboolean gst_base_audio_decoder_sink_event (GstPad * pad,
+    GstEvent * event);
+static gboolean gst_base_audio_decoder_src_event (GstPad * pad,
+    GstEvent * event);
+static GstFlowReturn gst_base_audio_decoder_chain (GstPad * pad,
+    GstBuffer * buf);
+static gboolean gst_base_audio_decoder_sink_query (GstPad * pad,
+    GstQuery * query);
+static GstStateChangeReturn gst_base_audio_decoder_change_state (GstElement *
+    element, GstStateChange transition);
+static const GstQueryType *gst_base_audio_decoder_get_query_types (GstPad *
+    pad);
+static gboolean gst_base_audio_decoder_src_query (GstPad * pad,
+    GstQuery * query);
+static gboolean gst_base_audio_decoder_src_convert (GstPad * pad,
+    GstFormat src_format, gint64 src_value, GstFormat * dest_format,
+    gint64 * dest_value);
+static void gst_base_audio_decoder_reset (GstBaseAudioDecoder *
+    base_audio_decoder);
+
+static guint64
+gst_base_audio_decoder_get_timestamp (GstBaseAudioDecoder * base_audio_decoder,
+    int picture_number);
+static guint64
+gst_base_audio_decoder_get_field_timestamp (GstBaseAudioDecoder *
+    base_audio_decoder, int field_offset);
+static GstAudioFrame *gst_base_audio_decoder_new_frame (GstBaseAudioDecoder *
+    base_audio_decoder);
+static void gst_base_audio_decoder_free_frame (GstAudioFrame * frame);
+
+GST_BOILERPLATE (GstBaseAudioDecoder, gst_base_audio_decoder,
+    GstBaseAudioCodec, GST_TYPE_BASE_AUDIO_CODEC);
+
+static void
+gst_base_audio_decoder_base_init (gpointer g_class)
+{
+
+}
+
+static void
+gst_base_audio_decoder_class_init (GstBaseAudioDecoderClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+
+  gobject_class = G_OBJECT_CLASS (klass);
+  gstelement_class = GST_ELEMENT_CLASS (klass);
+
+  gobject_class->finalize = gst_base_audio_decoder_finalize;
+
+  gstelement_class->change_state = gst_base_audio_decoder_change_state;
+
+  parent_class = g_type_class_peek_parent (klass);
+}
+
+static void
+gst_base_audio_decoder_init (GstBaseAudioDecoder * base_audio_decoder,
+    GstBaseAudioDecoderClass * klass)
+{
+  GstPad *pad;
+
+  GST_DEBUG ("gst_base_audio_decoder_init");
+
+  pad = GST_BASE_AUDIO_CODEC_SINK_PAD (base_audio_decoder);
+
+  gst_pad_set_chain_function (pad, gst_base_audio_decoder_chain);
+  gst_pad_set_event_function (pad, gst_base_audio_decoder_sink_event);
+  gst_pad_set_setcaps_function (pad, gst_base_audio_decoder_sink_setcaps);
+  gst_pad_set_query_function (pad, gst_base_audio_decoder_sink_query);
+
+  pad = GST_BASE_AUDIO_CODEC_SRC_PAD (base_audio_decoder);
+
+  gst_pad_set_event_function (pad, gst_base_audio_decoder_src_event);
+  gst_pad_set_query_type_function (pad, gst_base_audio_decoder_get_query_types);
+  gst_pad_set_query_function (pad, gst_base_audio_decoder_src_query);
+
+  base_audio_decoder->input_adapter = gst_adapter_new ();
+  base_audio_decoder->output_adapter = gst_adapter_new ();
+
+  gst_segment_init (&base_audio_decoder->state.segment, GST_FORMAT_TIME);
+  gst_base_audio_decoder_reset (base_audio_decoder);
+
+  base_audio_decoder->current_frame =
+      gst_base_audio_decoder_new_frame (base_audio_decoder);
+
+  base_audio_decoder->sink_clipping = TRUE;
+}
+
+static gboolean
+gst_base_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps)
+{
+  GstBaseAudioDecoder *base_audio_decoder;
+  GstBaseAudioDecoderClass *base_audio_decoder_class;
+  GstStructure *structure;
+  const GValue *codec_data;
+
+  base_audio_decoder = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+  base_audio_decoder_class =
+      GST_BASE_AUDIO_DECODER_GET_CLASS (base_audio_decoder);
+
+  GST_DEBUG ("setcaps %" GST_PTR_FORMAT, caps);
+
+  if (base_audio_decoder->codec_data) {
+    gst_buffer_unref (base_audio_decoder->codec_data);
+    base_audio_decoder->codec_data = NULL;
+  }
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  codec_data = gst_structure_get_value (structure, "codec_data");
+  if (codec_data && G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) {
+    base_audio_decoder->codec_data = gst_value_get_buffer (codec_data);
+  }
+
+  if (base_audio_decoder_class->start) {
+    base_audio_decoder_class->start (base_audio_decoder);
+  }
+
+  g_object_unref (base_audio_decoder);
+
+  return TRUE;
+}
+
+static void
+gst_base_audio_decoder_finalize (GObject * object)
+{
+  GstBaseAudioDecoder *base_audio_decoder;
+  GstBaseAudioDecoderClass *base_audio_decoder_class;
+
+  g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (object));
+  base_audio_decoder = GST_BASE_AUDIO_DECODER (object);
+  base_audio_decoder_class = GST_BASE_AUDIO_DECODER_GET_CLASS (object);
+
+  gst_base_audio_decoder_reset (base_audio_decoder);
+
+  GST_DEBUG_OBJECT (object, "finalize");
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static gboolean
+gst_base_audio_decoder_sink_event (GstPad * pad, GstEvent * event)
+{
+  GstBaseAudioDecoder *base_audio_decoder;
+  GstBaseAudioDecoderClass *base_audio_decoder_class;
+  gboolean ret = FALSE;
+
+  base_audio_decoder = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+  base_audio_decoder_class =
+      GST_BASE_AUDIO_DECODER_GET_CLASS (base_audio_decoder);
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_EOS:
+    {
+      GstAudioFrame *frame;
+
+      frame = g_malloc0 (sizeof (GstAudioFrame));
+      frame->presentation_frame_number =
+          base_audio_decoder->presentation_frame_number;
+      frame->presentation_duration = 0;
+      base_audio_decoder->presentation_frame_number++;
+
+      base_audio_decoder->frames =
+          g_list_append (base_audio_decoder->frames, frame);
+      if (base_audio_decoder_class->finish) {
+        base_audio_decoder_class->finish (base_audio_decoder, frame);
+      }
+
+      ret =
+          gst_pad_push_event (GST_BASE_AUDIO_CODEC_SRC_PAD (base_audio_decoder),
+          event);
+    }
+      break;
+    case GST_EVENT_NEWSEGMENT:
+    {
+      gboolean update;
+      double rate;
+      double applied_rate;
+      GstFormat format;
+      gint64 start;
+      gint64 stop;
+      gint64 position;
+
+      gst_event_parse_new_segment_full (event, &update, &rate,
+          &applied_rate, &format, &start, &stop, &position);
+
+      if (format != GST_FORMAT_TIME)
+        goto newseg_wrong_format;
+
+      GST_DEBUG ("new segment %lld %lld", start, position);
+
+      gst_segment_set_newsegment_full (&base_audio_decoder->state.segment,
+          update, rate, applied_rate, format, start, stop, position);
+
+      ret =
+          gst_pad_push_event (GST_BASE_AUDIO_CODEC_SRC_PAD (base_audio_decoder),
+          event);
+    }
+      break;
+    default:
+      /* FIXME this changes the order of events */
+      ret =
+          gst_pad_push_event (GST_BASE_AUDIO_CODEC_SRC_PAD (base_audio_decoder),
+          event);
+      break;
+  }
+
+done:
+  gst_object_unref (base_audio_decoder);
+  return ret;
+
+newseg_wrong_format:
+  {
+    GST_DEBUG_OBJECT (base_audio_decoder, "received non TIME newsegment");
+    gst_event_unref (event);
+    goto done;
+  }
+}
+
+static gboolean
+gst_base_audio_decoder_src_event (GstPad * pad, GstEvent * event)
+{
+  GstBaseAudioDecoder *base_audio_decoder;
+  gboolean res = FALSE;
+
+  base_audio_decoder = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_SEEK:
+    {
+      GstFormat format, tformat;
+      gdouble rate;
+      GstEvent *real_seek;
+      GstSeekFlags flags;
+      GstSeekType cur_type, stop_type;
+      gint64 cur, stop;
+      gint64 tcur, tstop;
+
+      gst_event_parse_seek (event, &rate, &format, &flags, &cur_type,
+          &cur, &stop_type, &stop);
+      gst_event_unref (event);
+
+      tformat = GST_FORMAT_TIME;
+      res =
+          gst_base_audio_decoder_src_convert (pad, format, cur, &tformat,
+          &tcur);
+      if (!res)
+        goto convert_error;
+      res =
+          gst_base_audio_decoder_src_convert (pad, format, stop, &tformat,
+          &tstop);
+      if (!res)
+        goto convert_error;
+
+      real_seek = gst_event_new_seek (rate, GST_FORMAT_TIME,
+          flags, cur_type, tcur, stop_type, tstop);
+
+      res =
+          gst_pad_push_event (GST_BASE_AUDIO_CODEC_SINK_PAD
+          (base_audio_decoder), real_seek);
+
+      break;
+    }
+    case GST_EVENT_QOS:
+    {
+      gdouble proportion;
+      GstClockTimeDiff diff;
+      GstClockTime timestamp;
+
+      gst_event_parse_qos (event, &proportion, &diff, &timestamp);
+
+      GST_OBJECT_LOCK (base_audio_decoder);
+      base_audio_decoder->proportion = proportion;
+      base_audio_decoder->earliest_time = timestamp + diff;
+      GST_OBJECT_UNLOCK (base_audio_decoder);
+
+      GST_DEBUG_OBJECT (base_audio_decoder,
+          "got QoS %" GST_TIME_FORMAT ", %" G_GINT64_FORMAT ", %g",
+          GST_TIME_ARGS (timestamp), diff, proportion);
+
+      res =
+          gst_pad_push_event (GST_BASE_AUDIO_CODEC_SINK_PAD
+          (base_audio_decoder), event);
+      break;
+    }
+    default:
+      res =
+          gst_pad_push_event (GST_BASE_AUDIO_CODEC_SINK_PAD
+          (base_audio_decoder), event);
+      break;
+  }
+done:
+  gst_object_unref (base_audio_decoder);
+  return res;
+
+convert_error:
+  GST_DEBUG_OBJECT (base_audio_decoder, "could not convert format");
+  goto done;
+}
+
+
+#if 0
+static gboolean
+gst_base_audio_decoder_sink_convert (GstPad * pad,
+    GstFormat src_format, gint64 src_value,
+    GstFormat * dest_format, gint64 * dest_value)
+{
+  gboolean res = TRUE;
+  GstBaseAudioDecoder *enc;
+
+  if (src_format == *dest_format) {
+    *dest_value = src_value;
+    return TRUE;
+  }
+
+  enc = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+
+  /* FIXME: check if we are in a decoding state */
+
+  switch (src_format) {
+    case GST_FORMAT_BYTES:
+      switch (*dest_format) {
+#if 0
+        case GST_FORMAT_DEFAULT:
+          *dest_value = gst_util_uint64_scale_int (src_value, 1,
+              enc->bytes_per_picture);
+          break;
+#endif
+        case GST_FORMAT_TIME:
+          /* seems like a rather silly conversion, implement me if you like */
+        default:
+          res = FALSE;
+      }
+      break;
+    case GST_FORMAT_DEFAULT:
+      switch (*dest_format) {
+        case GST_FORMAT_TIME:
+          *dest_value = gst_util_uint64_scale (src_value,
+              GST_SECOND * enc->fps_d, enc->fps_n);
+          break;
+#if 0
+        case GST_FORMAT_BYTES:
+          *dest_value = gst_util_uint64_scale_int (src_value,
+              enc->bytes_per_picture, 1);
+          break;
+#endif
+        default:
+          res = FALSE;
+      }
+      break;
+    default:
+      res = FALSE;
+      break;
+  }
+}
+#endif
+
+static gboolean
+gst_base_audio_decoder_src_convert (GstPad * pad,
+    GstFormat src_format, gint64 src_value,
+    GstFormat * dest_format, gint64 * dest_value)
+{
+  gboolean res = TRUE;
+  GstBaseAudioDecoder *enc;
+
+  if (src_format == *dest_format) {
+    *dest_value = src_value;
+    return TRUE;
+  }
+
+  enc = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+
+  /* FIXME: check if we are in a encoding state */
+
+  GST_DEBUG ("src convert");
+  switch (src_format) {
+#if 0
+    case GST_FORMAT_DEFAULT:
+      switch (*dest_format) {
+        case GST_FORMAT_TIME:
+          *dest_value = gst_util_uint64_scale (granulepos_to_frame (src_value),
+              enc->fps_d * GST_SECOND, enc->fps_n);
+          break;
+        default:
+          res = FALSE;
+      }
+      break;
+    case GST_FORMAT_TIME:
+      switch (*dest_format) {
+        case GST_FORMAT_DEFAULT:
+        {
+          *dest_value = gst_util_uint64_scale (src_value,
+              enc->fps_n, enc->fps_d * GST_SECOND);
+          break;
+        }
+        default:
+          res = FALSE;
+          break;
+      }
+      break;
+#endif
+    default:
+      res = FALSE;
+      break;
+  }
+
+  gst_object_unref (enc);
+
+  return res;
+}
+
+static const GstQueryType *
+gst_base_audio_decoder_get_query_types (GstPad * pad)
+{
+  static const GstQueryType query_types[] = {
+    GST_QUERY_CONVERT,
+    0
+  };
+
+  return query_types;
+}
+
+static gboolean
+gst_base_audio_decoder_src_query (GstPad * pad, GstQuery * query)
+{
+  GstBaseAudioDecoder *enc;
+  gboolean res;
+
+  enc = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+
+  switch GST_QUERY_TYPE
+    (query) {
+    case GST_QUERY_CONVERT:
+    {
+      GstFormat src_fmt, dest_fmt;
+      gint64 src_val, dest_val;
+
+      gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
+      res =
+          gst_base_audio_decoder_src_convert (pad, src_fmt, src_val, &dest_fmt,
+          &dest_val);
+      if (!res)
+        goto error;
+      gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
+      break;
+    }
+    default:
+      res = gst_pad_query_default (pad, query);
+    }
+  gst_object_unref (enc);
+  return res;
+
+error:
+  GST_DEBUG_OBJECT (enc, "query failed");
+  gst_object_unref (enc);
+  return res;
+}
+
+static gboolean
+gst_base_audio_decoder_sink_query (GstPad * pad, GstQuery * query)
+{
+  GstBaseAudioDecoder *base_audio_decoder;
+  gboolean res = FALSE;
+
+  base_audio_decoder = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+
+  GST_DEBUG_OBJECT (base_audio_decoder, "sink query fps=%d/%d",
+      base_audio_decoder->state.fps_n, base_audio_decoder->state.fps_d);
+  switch (GST_QUERY_TYPE (query)) {
+    case GST_QUERY_CONVERT:
+    {
+      GstFormat src_fmt, dest_fmt;
+      gint64 src_val, dest_val;
+
+      gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
+      res = gst_base_audio_rawaudio_convert (&base_audio_decoder->state,
+          src_fmt, src_val, &dest_fmt, &dest_val);
+      if (!res)
+        goto error;
+      gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
+      break;
+    }
+    default:
+      res = gst_pad_query_default (pad, query);
+      break;
+  }
+done:
+  gst_object_unref (base_audio_decoder);
+
+  return res;
+error:
+  GST_DEBUG_OBJECT (base_audio_decoder, "query failed");
+  goto done;
+}
+
+
+#if 0
+static gboolean
+gst_pad_is_negotiated (GstPad * pad)
+{
+  GstCaps *caps;
+
+  g_return_val_if_fail (pad != NULL, FALSE);
+
+  caps = gst_pad_get_negotiated_caps (pad);
+  if (caps) {
+    gst_caps_unref (caps);
+    return TRUE;
+  }
+
+  return FALSE;
+}
+#endif
+
+static void
+gst_base_audio_decoder_reset (GstBaseAudioDecoder * base_audio_decoder)
+{
+  GstBaseAudioDecoderClass *base_audio_decoder_class;
+  GList *g;
+
+  base_audio_decoder_class =
+      GST_BASE_AUDIO_DECODER_GET_CLASS (base_audio_decoder);
+
+  GST_DEBUG ("reset");
+
+  base_audio_decoder->started = FALSE;
+
+  base_audio_decoder->discont = TRUE;
+  base_audio_decoder->have_sync = FALSE;
+
+  base_audio_decoder->timestamp_offset = GST_CLOCK_TIME_NONE;
+  base_audio_decoder->system_frame_number = 0;
+  base_audio_decoder->presentation_frame_number = 0;
+  base_audio_decoder->last_sink_timestamp = GST_CLOCK_TIME_NONE;
+  base_audio_decoder->last_sink_offset_end = GST_CLOCK_TIME_NONE;
+  base_audio_decoder->base_picture_number = 0;
+  base_audio_decoder->last_timestamp = GST_CLOCK_TIME_NONE;
+
+  base_audio_decoder->offset = 0;
+
+  if (base_audio_decoder->caps) {
+    gst_caps_unref (base_audio_decoder->caps);
+    base_audio_decoder->caps = NULL;
+  }
+
+  if (base_audio_decoder->current_frame) {
+    gst_base_audio_decoder_free_frame (base_audio_decoder->current_frame);
+    base_audio_decoder->current_frame = NULL;
+  }
+
+  base_audio_decoder->have_src_caps = FALSE;
+
+  for (g = g_list_first (base_audio_decoder->frames); g; g = g_list_next (g)) {
+    GstAudioFrame *frame = g->data;
+    gst_base_audio_decoder_free_frame (frame);
+  }
+  g_list_free (base_audio_decoder->frames);
+  base_audio_decoder->frames = NULL;
+
+  if (base_audio_decoder_class->reset) {
+    base_audio_decoder_class->reset (base_audio_decoder);
+  }
+}
+
+static GstBuffer *
+gst_adapter_get_buffer (GstAdapter * adapter)
+{
+  return gst_buffer_ref (GST_BUFFER (adapter->buflist->data));
+
+}
+
+static GstFlowReturn
+gst_base_audio_decoder_chain (GstPad * pad, GstBuffer * buf)
+{
+  GstBaseAudioDecoder *base_audio_decoder;
+  GstBaseAudioDecoderClass *klass;
+  GstBuffer *buffer;
+  GstFlowReturn ret;
+
+  GST_DEBUG ("chain %lld", GST_BUFFER_TIMESTAMP (buf));
+
+#if 0
+  /* requiring the pad to be negotiated makes it impossible to use
+   * oggdemux or filesrc ! decoder */
+  if (!gst_pad_is_negotiated (pad)) {
+    GST_DEBUG ("not negotiated");
+    return GST_FLOW_NOT_NEGOTIATED;
+  }
+#endif
+
+  base_audio_decoder = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+  klass = GST_BASE_AUDIO_DECODER_GET_CLASS (base_audio_decoder);
+
+  GST_DEBUG_OBJECT (base_audio_decoder, "chain");
+
+  if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
+    GST_DEBUG_OBJECT (base_audio_decoder, "received DISCONT buffer");
+    if (base_audio_decoder->started) {
+      gst_base_audio_decoder_reset (base_audio_decoder);
+    }
+  }
+
+  if (!base_audio_decoder->started) {
+    klass->start (base_audio_decoder);
+    base_audio_decoder->started = TRUE;
+  }
+
+  if (GST_BUFFER_TIMESTAMP (buf) != GST_CLOCK_TIME_NONE) {
+    GST_DEBUG ("timestamp %lld offset %lld", GST_BUFFER_TIMESTAMP (buf),
+        base_audio_decoder->offset);
+    base_audio_decoder->last_sink_timestamp = GST_BUFFER_TIMESTAMP (buf);
+  }
+  if (GST_BUFFER_OFFSET_END (buf) != -1) {
+    GST_DEBUG ("gp %lld", GST_BUFFER_OFFSET_END (buf));
+    base_audio_decoder->last_sink_offset_end = GST_BUFFER_OFFSET_END (buf);
+  }
+  base_audio_decoder->offset += GST_BUFFER_SIZE (buf);
+
+#if 0
+  if (base_audio_decoder->timestamp_offset == GST_CLOCK_TIME_NONE &&
+      GST_BUFFER_TIMESTAMP (buf) != GST_CLOCK_TIME_NONE) {
+    GST_DEBUG ("got new offset %lld", GST_BUFFER_TIMESTAMP (buf));
+    base_audio_decoder->timestamp_offset = GST_BUFFER_TIMESTAMP (buf);
+  }
+#endif
+
+  if (base_audio_decoder->current_frame == NULL) {
+    base_audio_decoder->current_frame =
+        gst_base_audio_decoder_new_frame (base_audio_decoder);
+  }
+
+  gst_adapter_push (base_audio_decoder->input_adapter, buf);
+
+  if (!base_audio_decoder->have_sync) {
+    int n, m;
+
+    GST_DEBUG ("no sync, scanning");
+
+    n = gst_adapter_available (base_audio_decoder->input_adapter);
+    m = klass->scan_for_sync (base_audio_decoder, FALSE, 0, n);
+
+    if (m >= n) {
+      g_warning ("subclass scanned past end %d >= %d", m, n);
+    }
+
+    gst_adapter_flush (base_audio_decoder->input_adapter, m);
+
+    if (m < n) {
+      GST_DEBUG ("found possible sync after %d bytes (of %d)", m, n);
+
+      /* this is only "maybe" sync */
+      base_audio_decoder->have_sync = TRUE;
+    }
+
+    if (!base_audio_decoder->have_sync) {
+      gst_object_unref (base_audio_decoder);
+      return GST_FLOW_OK;
+    }
+  }
+
+  /* FIXME: use gst_adapter_prev_timestamp() here instead? */
+  buffer = gst_adapter_get_buffer (base_audio_decoder->input_adapter);
+
+  base_audio_decoder->buffer_timestamp = GST_BUFFER_TIMESTAMP (buffer);
+  gst_buffer_unref (buffer);
+
+  do {
+    ret = klass->parse_data (base_audio_decoder, FALSE);
+  } while (ret == GST_FLOW_OK);
+
+  if (ret == GST_BASE_AUDIO_DECODER_FLOW_NEED_DATA) {
+    gst_object_unref (base_audio_decoder);
+    return GST_FLOW_OK;
+  }
+
+  gst_object_unref (base_audio_decoder);
+  return ret;
+}
+
+static GstStateChangeReturn
+gst_base_audio_decoder_change_state (GstElement * element,
+    GstStateChange transition)
+{
+  GstBaseAudioDecoder *base_audio_decoder;
+  GstBaseAudioDecoderClass *base_audio_decoder_class;
+  GstStateChangeReturn ret;
+
+  base_audio_decoder = GST_BASE_AUDIO_DECODER (element);
+  base_audio_decoder_class = GST_BASE_AUDIO_DECODER_GET_CLASS (element);
+
+  switch (transition) {
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      if (base_audio_decoder_class->stop) {
+        base_audio_decoder_class->stop (base_audio_decoder);
+      }
+      break;
+    default:
+      break;
+  }
+
+  return ret;
+}
+
+static void
+gst_base_audio_decoder_free_frame (GstAudioFrame * frame)
+{
+  g_return_if_fail (frame != NULL);
+
+  if (frame->sink_buffer) {
+    gst_buffer_unref (frame->sink_buffer);
+  }
+#if 0
+  if (frame->src_buffer) {
+    gst_buffer_unref (frame->src_buffer);
+  }
+#endif
+
+  g_free (frame);
+}
+
+static GstAudioFrame *
+gst_base_audio_decoder_new_frame (GstBaseAudioDecoder * base_audio_decoder)
+{
+  GstAudioFrame *frame;
+
+  frame = g_malloc0 (sizeof (GstAudioFrame));
+
+  frame->system_frame_number = base_audio_decoder->system_frame_number;
+  base_audio_decoder->system_frame_number++;
+
+  frame->decode_frame_number = frame->system_frame_number -
+      base_audio_decoder->reorder_depth;
+
+  frame->decode_timestamp = -1;
+  frame->presentation_timestamp = -1;
+  frame->presentation_duration = -1;
+  frame->n_fields = 2;
+
+  return frame;
+}
+
+GstFlowReturn
+gst_base_audio_decoder_finish_frame (GstBaseAudioDecoder * base_audio_decoder,
+    GstAudioFrame * frame)
+{
+  GstBaseAudioDecoderClass *base_audio_decoder_class;
+  GstBuffer *src_buffer;
+
+  GST_DEBUG ("finish frame");
+
+  base_audio_decoder_class =
+      GST_BASE_AUDIO_DECODER_GET_CLASS (base_audio_decoder);
+
+  GST_DEBUG ("finish frame sync=%d pts=%lld", frame->is_sync_point,
+      frame->presentation_timestamp);
+
+  if (frame->is_sync_point) {
+    if (GST_CLOCK_TIME_IS_VALID (frame->presentation_timestamp)) {
+      if (frame->presentation_timestamp != base_audio_decoder->timestamp_offset) {
+        GST_DEBUG ("sync timestamp %lld diff %lld",
+            frame->presentation_timestamp,
+            frame->presentation_timestamp -
+            base_audio_decoder->state.segment.start);
+        base_audio_decoder->timestamp_offset = frame->presentation_timestamp;
+        base_audio_decoder->field_index = 0;
+      } else {
+        /* This case is for one initial timestamp and no others, e.g.,
+         * filesrc ! decoder ! xvimagesink */
+        GST_WARNING ("sync timestamp didn't change, ignoring");
+        frame->presentation_timestamp = GST_CLOCK_TIME_NONE;
+      }
+    } else {
+      GST_WARNING ("sync point doesn't have timestamp");
+      if (GST_CLOCK_TIME_IS_VALID (base_audio_decoder->timestamp_offset)) {
+        GST_ERROR ("No base timestamp.  Assuming frames start at 0");
+        base_audio_decoder->timestamp_offset = 0;
+        base_audio_decoder->field_index = 0;
+      }
+    }
+  }
+  frame->field_index = base_audio_decoder->field_index;
+  base_audio_decoder->field_index += frame->n_fields;
+
+  if (frame->presentation_timestamp == GST_CLOCK_TIME_NONE) {
+    frame->presentation_timestamp =
+        gst_base_audio_decoder_get_field_timestamp (base_audio_decoder,
+        frame->field_index);
+    frame->presentation_duration = GST_CLOCK_TIME_NONE;
+    frame->decode_timestamp =
+        gst_base_audio_decoder_get_timestamp (base_audio_decoder,
+        frame->decode_frame_number);
+  }
+  if (frame->presentation_duration == GST_CLOCK_TIME_NONE) {
+    frame->presentation_duration =
+        gst_base_audio_decoder_get_field_timestamp (base_audio_decoder,
+        frame->field_index + frame->n_fields) - frame->presentation_timestamp;
+  }
+
+  if (GST_CLOCK_TIME_IS_VALID (base_audio_decoder->last_timestamp)) {
+    if (frame->presentation_timestamp < base_audio_decoder->last_timestamp) {
+      GST_WARNING ("decreasing timestamp (%lld < %lld)",
+          frame->presentation_timestamp, base_audio_decoder->last_timestamp);
+    }
+  }
+  base_audio_decoder->last_timestamp = frame->presentation_timestamp;
+
+  GST_BUFFER_FLAG_UNSET (frame->src_buffer, GST_BUFFER_FLAG_DELTA_UNIT);
+  if (base_audio_decoder->state.interlaced) {
+#ifndef GST_AUDIO_BUFFER_TFF
+#define GST_AUDIO_BUFFER_TFF (GST_MINI_OBJECT_FLAG_LAST << 5)
+#endif
+#ifndef GST_AUDIO_BUFFER_RFF
+#define GST_AUDIO_BUFFER_RFF (GST_MINI_OBJECT_FLAG_LAST << 6)
+#endif
+#ifndef GST_AUDIO_BUFFER_ONEFIELD
+#define GST_AUDIO_BUFFER_ONEFIELD (GST_MINI_OBJECT_FLAG_LAST << 7)
+#endif
+    int tff = base_audio_decoder->state.top_field_first;
+
+    if (frame->field_index & 1) {
+      tff ^= 1;
+    }
+    if (tff) {
+      GST_BUFFER_FLAG_SET (frame->src_buffer, GST_AUDIO_BUFFER_TFF);
+    } else {
+      GST_BUFFER_FLAG_UNSET (frame->src_buffer, GST_AUDIO_BUFFER_TFF);
+    }
+    GST_BUFFER_FLAG_UNSET (frame->src_buffer, GST_AUDIO_BUFFER_RFF);
+    GST_BUFFER_FLAG_UNSET (frame->src_buffer, GST_AUDIO_BUFFER_ONEFIELD);
+    if (frame->n_fields == 3) {
+      GST_BUFFER_FLAG_SET (frame->src_buffer, GST_AUDIO_BUFFER_RFF);
+    } else if (frame->n_fields == 1) {
+      GST_BUFFER_FLAG_UNSET (frame->src_buffer, GST_AUDIO_BUFFER_ONEFIELD);
+    }
+  }
+
+  GST_BUFFER_TIMESTAMP (frame->src_buffer) = frame->presentation_timestamp;
+  GST_BUFFER_DURATION (frame->src_buffer) = frame->presentation_duration;
+  GST_BUFFER_OFFSET (frame->src_buffer) = -1;
+  GST_BUFFER_OFFSET_END (frame->src_buffer) = -1;
+
+  GST_DEBUG ("pushing frame %lld", frame->presentation_timestamp);
+
+  base_audio_decoder->frames =
+      g_list_remove (base_audio_decoder->frames, frame);
+
+  gst_base_audio_decoder_set_src_caps (base_audio_decoder);
+
+  src_buffer = frame->src_buffer;
+  frame->src_buffer = NULL;
+
+  gst_base_audio_decoder_free_frame (frame);
+
+  if (base_audio_decoder->sink_clipping) {
+    gint64 start = GST_BUFFER_TIMESTAMP (src_buffer);
+    gint64 stop = GST_BUFFER_TIMESTAMP (src_buffer) +
+        GST_BUFFER_DURATION (src_buffer);
+
+    if (gst_segment_clip (&base_audio_decoder->state.segment, GST_FORMAT_TIME,
+            start, stop, &start, &stop)) {
+      GST_BUFFER_TIMESTAMP (src_buffer) = start;
+      GST_BUFFER_DURATION (src_buffer) = stop - start;
+    } else {
+      GST_DEBUG ("dropping buffer outside segment");
+      gst_buffer_unref (src_buffer);
+      return GST_FLOW_OK;
+    }
+  }
+
+  return gst_pad_push (GST_BASE_AUDIO_CODEC_SRC_PAD (base_audio_decoder),
+      src_buffer);
+}
+
+int
+gst_base_audio_decoder_get_height (GstBaseAudioDecoder * base_audio_decoder)
+{
+  return base_audio_decoder->state.height;
+}
+
+int
+gst_base_audio_decoder_get_width (GstBaseAudioDecoder * base_audio_decoder)
+{
+  return base_audio_decoder->state.width;
+}
+
+GstFlowReturn
+gst_base_audio_decoder_end_of_stream (GstBaseAudioDecoder * base_audio_decoder,
+    GstBuffer * buffer)
+{
+
+  if (base_audio_decoder->frames) {
+    GST_DEBUG ("EOS with frames left over");
+  }
+
+  return gst_pad_push (GST_BASE_AUDIO_CODEC_SRC_PAD (base_audio_decoder),
+      buffer);
+}
+
+void
+gst_base_audio_decoder_add_to_frame (GstBaseAudioDecoder * base_audio_decoder,
+    int n_bytes)
+{
+  GstBuffer *buf;
+
+  GST_DEBUG ("add to frame");
+
+#if 0
+  if (gst_adapter_available (base_audio_decoder->output_adapter) == 0) {
+    GstBuffer *buffer;
+
+    buffer =
+        gst_adapter_get_orig_buffer_at_offset
+        (base_audio_decoder->input_adapter, 0);
+    if (buffer) {
+      base_audio_decoder->current_frame->presentation_timestamp =
+          GST_BUFFER_TIMESTAMP (buffer);
+      gst_buffer_unref (buffer);
+    }
+  }
+#endif
+
+  if (n_bytes == 0)
+    return;
+
+  buf = gst_adapter_take_buffer (base_audio_decoder->input_adapter, n_bytes);
+
+  gst_adapter_push (base_audio_decoder->output_adapter, buf);
+}
+
+static guint64
+gst_base_audio_decoder_get_timestamp (GstBaseAudioDecoder * base_audio_decoder,
+    int picture_number)
+{
+  if (base_audio_decoder->state.fps_d == 0) {
+    return -1;
+  }
+  if (picture_number < base_audio_decoder->base_picture_number) {
+    return base_audio_decoder->timestamp_offset -
+        (gint64) gst_util_uint64_scale (base_audio_decoder->base_picture_number
+        - picture_number, base_audio_decoder->state.fps_d * GST_SECOND,
+        base_audio_decoder->state.fps_n);
+  } else {
+    return base_audio_decoder->timestamp_offset +
+        gst_util_uint64_scale (picture_number -
+        base_audio_decoder->base_picture_number,
+        base_audio_decoder->state.fps_d * GST_SECOND,
+        base_audio_decoder->state.fps_n);
+  }
+}
+
+static guint64
+gst_base_audio_decoder_get_field_timestamp (GstBaseAudioDecoder *
+    base_audio_decoder, int field_offset)
+{
+  if (base_audio_decoder->state.fps_d == 0) {
+    return GST_CLOCK_TIME_NONE;
+  }
+  if (field_offset < 0) {
+    GST_WARNING ("field offset < 0");
+    return GST_CLOCK_TIME_NONE;
+  }
+  return base_audio_decoder->timestamp_offset +
+      gst_util_uint64_scale (field_offset,
+      base_audio_decoder->state.fps_d * GST_SECOND,
+      base_audio_decoder->state.fps_n * 2);
+}
+
+
+GstFlowReturn
+gst_base_audio_decoder_have_frame (GstBaseAudioDecoder * base_audio_decoder)
+{
+  GstAudioFrame *frame = base_audio_decoder->current_frame;
+  GstBuffer *buffer;
+  GstBaseAudioDecoderClass *base_audio_decoder_class;
+  GstFlowReturn ret = GST_FLOW_OK;
+  int n_available;
+
+  GST_DEBUG ("have_frame");
+
+  base_audio_decoder_class =
+      GST_BASE_AUDIO_DECODER_GET_CLASS (base_audio_decoder);
+
+  n_available = gst_adapter_available (base_audio_decoder->output_adapter);
+  if (n_available) {
+    buffer = gst_adapter_take_buffer (base_audio_decoder->output_adapter,
+        n_available);
+  } else {
+    buffer = gst_buffer_new_and_alloc (0);
+  }
+
+  frame->distance_from_sync = base_audio_decoder->distance_from_sync;
+  base_audio_decoder->distance_from_sync++;
+
+#if 0
+  if (frame->presentation_timestamp == GST_CLOCK_TIME_NONE) {
+    frame->presentation_timestamp =
+        gst_base_audio_decoder_get_timestamp (base_audio_decoder,
+        frame->presentation_frame_number);
+    frame->presentation_duration =
+        gst_base_audio_decoder_get_timestamp (base_audio_decoder,
+        frame->presentation_frame_number + 1) - frame->presentation_timestamp;
+    frame->decode_timestamp =
+        gst_base_audio_decoder_get_timestamp (base_audio_decoder,
+        frame->decode_frame_number);
+  }
+#endif
+
+#if 0
+  GST_BUFFER_TIMESTAMP (buffer) = frame->presentation_timestamp;
+  GST_BUFFER_DURATION (buffer) = frame->presentation_duration;
+  if (frame->decode_frame_number < 0) {
+    GST_BUFFER_OFFSET (buffer) = 0;
+  } else {
+    GST_BUFFER_OFFSET (buffer) = frame->decode_timestamp;
+  }
+  GST_BUFFER_OFFSET_END (buffer) = GST_CLOCK_TIME_NONE;
+#endif
+
+  GST_DEBUG ("pts %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (frame->presentation_timestamp));
+  GST_DEBUG ("dts %" GST_TIME_FORMAT, GST_TIME_ARGS (frame->decode_timestamp));
+  GST_DEBUG ("dist %d", frame->distance_from_sync);
+
+  if (frame->is_sync_point) {
+    GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DELTA_UNIT);
+  } else {
+    GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DELTA_UNIT);
+  }
+  if (base_audio_decoder->discont) {
+    GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
+    base_audio_decoder->discont = FALSE;
+  }
+
+  frame->sink_buffer = buffer;
+
+  base_audio_decoder->frames = g_list_append (base_audio_decoder->frames,
+      frame);
+
+  /* do something with frame */
+  ret = base_audio_decoder_class->handle_frame (base_audio_decoder, frame);
+  if (!GST_FLOW_IS_SUCCESS (ret)) {
+    GST_DEBUG ("flow error!");
+  }
+
+  /* create new frame */
+  base_audio_decoder->current_frame =
+      gst_base_audio_decoder_new_frame (base_audio_decoder);
+
+  return ret;
+}
+
+GstAudioState *
+gst_base_audio_decoder_get_state (GstBaseAudioDecoder * base_audio_decoder)
+{
+  return &base_audio_decoder->state;
+
+}
+
+void
+gst_base_audio_decoder_set_state (GstBaseAudioDecoder * base_audio_decoder,
+    GstAudioState * state)
+{
+  memcpy (&base_audio_decoder->state, state, sizeof (*state));
+
+}
+
+void
+gst_base_audio_decoder_lost_sync (GstBaseAudioDecoder * base_audio_decoder)
+{
+  g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (base_audio_decoder));
+
+  GST_DEBUG ("lost_sync");
+
+  if (gst_adapter_available (base_audio_decoder->input_adapter) >= 1) {
+    gst_adapter_flush (base_audio_decoder->input_adapter, 1);
+  }
+
+  base_audio_decoder->have_sync = FALSE;
+}
+
+void
+gst_base_audio_decoder_set_sync_point (GstBaseAudioDecoder * base_audio_decoder)
+{
+  GST_DEBUG ("set_sync_point");
+
+  base_audio_decoder->current_frame->is_sync_point = TRUE;
+  base_audio_decoder->distance_from_sync = 0;
+
+  base_audio_decoder->current_frame->presentation_timestamp =
+      base_audio_decoder->last_sink_timestamp;
+
+
+}
+
+GstAudioFrame *
+gst_base_audio_decoder_get_frame (GstBaseAudioDecoder * base_audio_decoder,
+    int frame_number)
+{
+  GList *g;
+
+  for (g = g_list_first (base_audio_decoder->frames); g; g = g_list_next (g)) {
+    GstAudioFrame *frame = g->data;
+
+    if (frame->system_frame_number == frame_number) {
+      return frame;
+    }
+  }
+
+  return NULL;
+}
+
+void
+gst_base_audio_decoder_set_src_caps (GstBaseAudioDecoder * base_audio_decoder)
+{
+  GstCaps *caps;
+  GstAudioState *state = &base_audio_decoder->state;
+
+  if (base_audio_decoder->have_src_caps)
+    return;
+
+  caps = gst_audio_format_new_caps (state->format,
+      state->width, state->height,
+      state->fps_n, state->fps_d, state->par_n, state->par_d);
+  gst_caps_set_simple (caps, "interlaced",
+      G_TYPE_BOOLEAN, state->interlaced, NULL);
+
+  GST_DEBUG ("setting caps %" GST_PTR_FORMAT, caps);
+
+  gst_pad_set_caps (GST_BASE_AUDIO_CODEC_SRC_PAD (base_audio_decoder), caps);
+
+  base_audio_decoder->have_src_caps = TRUE;
+}
diff --git a/gst-libs/gst/audio/gstbaseaudiodecoder.h b/gst-libs/gst/audio/gstbaseaudiodecoder.h
new file mode 100644
index 0000000..cd9676c
--- /dev/null
+++ b/gst-libs/gst/audio/gstbaseaudiodecoder.h
@@ -0,0 +1,67 @@
+/* GStreamer
+ * Copyright (C) 2009 Igalia S.L.
+ * Author: Iago Toral Quiroga <itoral@igalia.com>
+ *
+ * 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 _GST_BASE_AUDIO_DECODER_H_
+#define _GST_BASE_AUDIO_DECODER_H_
+
+#ifndef GST_USE_UNSTABLE_API
+#warning "GstBaseAudioDecoder is unstable API and may change in future."
+#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
+#endif
+
+#include <gst/audio/gstbaseaudiocodec.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_BASE_AUDIO_DECODER \
+  (gst_base_audio_decoder_get_type())
+#define GST_BASE_AUDIO_DECODER(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoder))
+#define GST_BASE_AUDIO_DECODER_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoderClass))
+#define GST_BASE_AUDIO_DECODER_GET_CLASS(obj) \
+  (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoderClass))
+#define GST_IS_BASE_AUDIO_DECODER(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_DECODER))
+#define GST_IS_BASE_AUDIO_DECODER_CLASS(obj) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_DECODER))
+
+typedef struct _GstBaseAudioDecoder GstBaseAudioDecoder;
+typedef struct _GstBaseAudioDecoderClass GstBaseAudioDecoderClass;
+
+struct _GstBaseAudioDecoder
+{
+  GstBaseAudioCodec base_audio_codec;
+
+  /*< private >*/
+};
+
+struct _GstBaseAudioDecoderClass
+{
+  GstBaseAudioCodecClass base_audio_codec_class;
+
+};
+
+GType gst_base_audio_decoder_get_type (void);
+
+G_END_DECLS
+
+#endif
+