fdkaac: Add FDK AAC based decoder element

Missing is support for error concealment, etc.

ext/fdkaac/Makefile.am

@@ -1,6 +1,6 @@
 plugin_LTLIBRARIES = libgstfdkaac.la
 
-libgstfdkaac_la_SOURCES = plugin.c gstfdkaacenc.c
+libgstfdkaac_la_SOURCES = plugin.c gstfdkaacenc.c gstfdkaacdec.c
 libgstfdkaac_la_CFLAGS = -DGST_USE_UNSTABLE_API \
 	$(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) \
 	$(GST_CFLAGS) $(FDK_AAC_CFLAGS)
@@ -10,4 +10,4 @@ libgstfdkaac_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) \
 libgstfdkaac_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
 libgstfdkaac_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
 
-noinst_HEADERS = gstfdkaacenc.h
+noinst_HEADERS = gstfdkaacenc.h gstfdkaacdec.h

ext/fdkaac/gstfdkaacdec.c

@@ -0,0 +1,438 @@
+/*
+ * Copyright (C) 2016 Sebastian Dröge <sebastian@centricular.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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstfdkaacdec.h"
+
+#include <gst/pbutils/pbutils.h>
+
+#include <string.h>
+
+/* TODO:
+ * - LOAS / LATM support
+ * - Error concealment
+ */
+
+static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/mpeg, "
+        "mpegversion = (int) 4, "
+        "stream-format = (string) { adts, adif, raw }")
+    );
+
+static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) " GST_AUDIO_NE (S16) ", "
+        "layout = (string) interleaved, "
+        "rate = (int) [8000, 96000], " "channels = (int) [1, 8]")
+    );
+
+GST_DEBUG_CATEGORY_STATIC (gst_fdkaacdec_debug);
+#define GST_CAT_DEFAULT gst_fdkaacdec_debug
+
+static gboolean gst_fdkaacdec_start (GstAudioDecoder * dec);
+static gboolean gst_fdkaacdec_stop (GstAudioDecoder * dec);
+static gboolean gst_fdkaacdec_set_format (GstAudioDecoder * dec,
+    GstCaps * caps);
+static GstFlowReturn gst_fdkaacdec_handle_frame (GstAudioDecoder * dec,
+    GstBuffer * in_buf);
+static void gst_fdkaacdec_flush (GstAudioDecoder * dec, gboolean hard);
+
+G_DEFINE_TYPE (GstFdkAacDec, gst_fdkaacdec, GST_TYPE_AUDIO_DECODER);
+
+static gboolean
+gst_fdkaacdec_start (GstAudioDecoder * dec)
+{
+  GstFdkAacDec *self = GST_FDKAACDEC (dec);
+
+  GST_DEBUG_OBJECT (self, "start");
+
+  return TRUE;
+}
+
+static gboolean
+gst_fdkaacdec_stop (GstAudioDecoder * dec)
+{
+  GstFdkAacDec *self = GST_FDKAACDEC (dec);
+
+  GST_DEBUG_OBJECT (self, "stop");
+
+  g_free (self->decode_buffer);
+  self->decode_buffer = NULL;
+
+  if (self->dec)
+    aacDecoder_Close (self->dec);
+  self->dec = NULL;
+
+  return TRUE;
+}
+
+static gboolean
+gst_fdkaacdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
+{
+  GstFdkAacDec *self = GST_FDKAACDEC (dec);
+  TRANSPORT_TYPE transport_format;
+  GstStructure *s;
+  const gchar *stream_format;
+  AAC_DECODER_ERROR err;
+
+  if (self->dec) {
+    /* drain */
+    gst_fdkaacdec_handle_frame (dec, NULL);
+    aacDecoder_Close (self->dec);
+    self->dec = NULL;
+  }
+
+  s = gst_caps_get_structure (caps, 0);
+  stream_format = gst_structure_get_string (s, "stream-format");
+  if (strcmp (stream_format, "raw") == 0) {
+    transport_format = TT_MP4_RAW;
+  } else if (strcmp (stream_format, "adif") == 0) {
+    transport_format = TT_MP4_ADIF;
+  } else if (strcmp (stream_format, "adts") == 0) {
+    transport_format = TT_MP4_ADTS;
+  } else {
+    g_assert_not_reached ();
+  }
+
+  self->dec = aacDecoder_Open (transport_format, 1);
+  if (!self->dec) {
+    GST_ERROR_OBJECT (self, "Failed to open decoder");
+    return FALSE;
+  }
+
+  if (transport_format == TT_MP4_RAW) {
+    GstBuffer *codec_data = NULL;
+    GstMapInfo map;
+    guint8 *data;
+    guint size;
+
+    gst_structure_get (s, "codec_data", GST_TYPE_BUFFER, &codec_data, NULL);
+
+    if (!codec_data) {
+      GST_ERROR_OBJECT (self, "Raw AAC without codec_data not supported");
+      return FALSE;
+    }
+
+    gst_buffer_map (codec_data, &map, GST_MAP_READ);
+    data = map.data;
+    size = map.size;
+
+    if ((err = aacDecoder_ConfigRaw (self->dec, &data, &size)) != AAC_DEC_OK) {
+      gst_buffer_unmap (codec_data, &map);
+      GST_ERROR_OBJECT (self, "Invalid codec_data: %d", err);
+      return FALSE;
+    }
+
+    gst_buffer_unmap (codec_data, &map);
+  }
+
+  if ((err =
+          aacDecoder_SetParam (self->dec, AAC_PCM_OUTPUT_CHANNEL_MAPPING,
+              0)) != AAC_DEC_OK) {
+    GST_ERROR_OBJECT (self, "Failed to set output channel mapping: %d", err);
+    return FALSE;
+  }
+
+  if ((err =
+          aacDecoder_SetParam (self->dec, AAC_PCM_OUTPUT_INTERLEAVED,
+              0)) != AAC_DEC_OK) {
+    GST_ERROR_OBJECT (self, "Failed to set output channel mapping: %d", err);
+    return FALSE;
+  }
+
+  /* 8 channels * 2 bytes per sample * 2048 samples */
+  if (!self->decode_buffer) {
+    self->decode_buffer_size = 8 * 2 * 2048;
+    self->decode_buffer = g_malloc (self->decode_buffer_size);
+  }
+
+  return TRUE;
+}
+
+static GstFlowReturn
+gst_fdkaacdec_handle_frame (GstAudioDecoder * dec, GstBuffer * inbuf)
+{
+  GstFdkAacDec *self = GST_FDKAACDEC (dec);
+  GstFlowReturn ret = GST_FLOW_OK;
+  GstBuffer *outbuf;
+  GstMapInfo imap;
+  AAC_DECODER_ERROR err;
+  guint size, valid;
+  CStreamInfo *stream_info;
+  GstAudioInfo info;
+  guint flags = 0, i;
+  GstAudioChannelPosition pos[64], gst_pos[64];
+  gboolean need_reorder;
+
+  if (inbuf) {
+    gst_buffer_map (inbuf, &imap, GST_MAP_READ);
+    valid = size = imap.size;
+
+    if ((err =
+            aacDecoder_Fill (self->dec, (guint8 **) & imap.data, &size,
+                &valid)) != AAC_DEC_OK) {
+      GST_AUDIO_DECODER_ERROR (self, 1, STREAM, DECODE, (NULL),
+          ("filling error: %d", err), ret);
+      gst_buffer_unmap (inbuf, &imap);
+      goto out;
+    }
+    gst_buffer_unmap (inbuf, &imap);
+
+    if (GST_BUFFER_IS_DISCONT (inbuf))
+      flags |= AACDEC_INTR;
+  } else {
+    flags |= AACDEC_FLUSH;
+  }
+
+  if ((err =
+          aacDecoder_DecodeFrame (self->dec, (gint16 *) self->decode_buffer,
+              self->decode_buffer_size, flags)) != AAC_DEC_OK) {
+    GST_AUDIO_DECODER_ERROR (self, 1, STREAM, DECODE, (NULL),
+        ("decoding error: %d", err), ret);
+    goto out;
+  }
+
+  stream_info = aacDecoder_GetStreamInfo (self->dec);
+  if (!stream_info) {
+    GST_AUDIO_DECODER_ERROR (self, 1, STREAM, DECODE, (NULL),
+        ("failed to get stream info"), ret);
+    goto out;
+  }
+
+  /* FIXME: Don't recalculate this on every buffer */
+  if (stream_info->numChannels == 1) {
+    pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
+  } else {
+    gint n_front = 0, n_side = 0, n_back = 0, n_lfe = 0;
+
+    /* FIXME: Can this be simplified somehow? */
+    for (i = 0; i < stream_info->numChannels; i++) {
+      if (stream_info->pChannelType[i] == ACT_FRONT) {
+        n_front++;
+      } else if (stream_info->pChannelType[i] == ACT_SIDE) {
+        n_side++;
+      } else if (stream_info->pChannelType[i] == ACT_BACK) {
+        n_back++;
+      } else if (stream_info->pChannelType[i] == ACT_LFE) {
+        n_lfe++;
+      } else {
+        GST_ERROR_OBJECT (self, "Channel type %d not supported",
+            stream_info->pChannelType[i]);
+        ret = GST_FLOW_NOT_NEGOTIATED;
+        goto out;
+      }
+    }
+
+    for (i = 0; i < stream_info->numChannels; i++) {
+      if (stream_info->pChannelType[i] == ACT_FRONT) {
+        if (stream_info->pChannelIndices[i] == 0) {
+          if (n_front & 1)
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
+          else if (n_front > 2)
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER;
+          else
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+        } else if (stream_info->pChannelIndices[i] == 1) {
+          if ((n_front & 1) && n_front > 3)
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER;
+          else if (n_front & 1)
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+          else if (n_front > 2)
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER;
+          else
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+        } else if (stream_info->pChannelIndices[i] == 2) {
+          if ((n_front & 1) && n_front > 3)
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER;
+          else if (n_front & 1)
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+          else if (n_front > 2)
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+          else
+            g_assert_not_reached ();
+        } else if (stream_info->pChannelIndices[i] == 3) {
+          if ((n_front & 1) && n_front > 3)
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+          else if (n_front & 1)
+            g_assert_not_reached ();
+          else if (n_front > 2)
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+          else
+            g_assert_not_reached ();
+        } else if (stream_info->pChannelIndices[i] == 4) {
+          if ((n_front & 1) && n_front > 2)
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+          else if (n_front & 1)
+            g_assert_not_reached ();
+          else if (n_front > 2)
+            g_assert_not_reached ();
+          else
+            g_assert_not_reached ();
+        } else {
+          GST_ERROR_OBJECT (self, "Front channel index %d not supported",
+              stream_info->pChannelIndices[i]);
+          ret = GST_FLOW_NOT_NEGOTIATED;
+          goto out;
+        }
+      } else if (stream_info->pChannelType[i] == ACT_SIDE) {
+        if (n_side & 1) {
+          GST_ERROR_OBJECT (self, "Odd number of side channels not supported");
+          ret = GST_FLOW_NOT_NEGOTIATED;
+          goto out;
+        } else if (stream_info->pChannelIndices[i] == 0) {
+          pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
+        } else if (stream_info->pChannelIndices[i] == 1) {
+          pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
+        } else {
+          GST_ERROR_OBJECT (self, "Side channel index %d not supported",
+              stream_info->pChannelIndices[i]);
+          ret = GST_FLOW_NOT_NEGOTIATED;
+          goto out;
+        }
+      } else if (stream_info->pChannelType[i] == ACT_BACK) {
+        if (stream_info->pChannelIndices[i] == 0) {
+          if (n_back & 1)
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
+          else
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
+        } else if (stream_info->pChannelIndices[i] == 1) {
+          if (n_back & 1)
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
+          else
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
+        } else if (stream_info->pChannelIndices[i] == 2) {
+          if (n_back & 1)
+            pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
+          else
+            g_assert_not_reached ();
+        } else {
+          GST_ERROR_OBJECT (self, "Side channel index %d not supported",
+              stream_info->pChannelIndices[i]);
+          ret = GST_FLOW_NOT_NEGOTIATED;
+          goto out;
+        }
+      } else if (stream_info->pChannelType[i] == ACT_LFE) {
+        if (stream_info->pChannelIndices[i] == 0) {
+          pos[i] = GST_AUDIO_CHANNEL_POSITION_LFE1;
+        } else {
+          GST_ERROR_OBJECT (self, "LFE channel index %d not supported",
+              stream_info->pChannelIndices[i]);
+          ret = GST_FLOW_NOT_NEGOTIATED;
+          goto out;
+        }
+      } else {
+        GST_ERROR_OBJECT (self, "Channel type %d not supported",
+            stream_info->pChannelType[i]);
+        ret = GST_FLOW_NOT_NEGOTIATED;
+        goto out;
+      }
+    }
+  }
+
+  memcpy (gst_pos, pos,
+      sizeof (GstAudioChannelPosition) * stream_info->numChannels);
+  if (!gst_audio_channel_positions_to_valid_order (gst_pos,
+          stream_info->numChannels)) {
+    ret = GST_FLOW_NOT_NEGOTIATED;
+    goto out;
+  }
+
+  need_reorder =
+      memcmp (pos, gst_pos,
+      sizeof (GstAudioChannelPosition) * stream_info->numChannels) != 0;
+
+  gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16,
+      stream_info->sampleRate, stream_info->numChannels, gst_pos);
+  if (!gst_audio_decoder_set_output_format (dec, &info)) {
+    GST_ERROR_OBJECT (self, "Failed to set output format");
+    ret = GST_FLOW_NOT_NEGOTIATED;
+    goto out;
+  }
+
+  outbuf =
+      gst_audio_decoder_allocate_output_buffer (dec,
+      stream_info->frameSize * GST_AUDIO_INFO_BPF (&info));
+  gst_buffer_fill (outbuf, 0, self->decode_buffer,
+      gst_buffer_get_size (outbuf));
+
+  if (need_reorder) {
+    gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_INFO_FORMAT (&info),
+        GST_AUDIO_INFO_CHANNELS (&info), pos, gst_pos);
+  }
+
+  ret = gst_audio_decoder_finish_frame (dec, outbuf, 1);
+
+out:
+
+  return ret;
+}
+
+static void
+gst_fdkaacdec_flush (GstAudioDecoder * dec, gboolean hard)
+{
+  GstFdkAacDec *self = GST_FDKAACDEC (dec);
+
+  if (self->dec) {
+    AAC_DECODER_ERROR err;
+    if ((err =
+            aacDecoder_DecodeFrame (self->dec, (gint16 *) self->decode_buffer,
+                self->decode_buffer_size, AACDEC_FLUSH)) != AAC_DEC_OK) {
+      GST_ERROR_OBJECT (self, "flushing error: %d", err);
+    }
+  }
+}
+
+static void
+gst_fdkaacdec_init (GstFdkAacDec * self)
+{
+  self->dec = NULL;
+
+  gst_audio_decoder_set_drainable (GST_AUDIO_DECODER (self), TRUE);
+  gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (self), TRUE);
+}
+
+static void
+gst_fdkaacdec_class_init (GstFdkAacDecClass * klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
+
+  base_class->start = GST_DEBUG_FUNCPTR (gst_fdkaacdec_start);
+  base_class->stop = GST_DEBUG_FUNCPTR (gst_fdkaacdec_stop);
+  base_class->set_format = GST_DEBUG_FUNCPTR (gst_fdkaacdec_set_format);
+  base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_fdkaacdec_handle_frame);
+  base_class->flush = GST_DEBUG_FUNCPTR (gst_fdkaacdec_flush);
+
+  gst_element_class_add_static_pad_template (element_class, &sink_template);
+  gst_element_class_add_static_pad_template (element_class, &src_template);
+
+  gst_element_class_set_static_metadata (element_class, "FDK AAC audio decoder",
+      "Codec/Decoder/Audio", "FDK AAC audio decoder",
+      "Sebastian Dröge <sebastian@centricular.com>");
+
+  GST_DEBUG_CATEGORY_INIT (gst_fdkaacdec_debug, "fdkaacdec", 0,
+      "fdkaac decoder");
+}

ext/fdkaac/gstfdkaacdec.h

@@ -0,0 +1,60 @@
+/*
+ * Copyright (C) 2016 Sebastian Dröge <sebastian@centricular.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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_FDKAACDEC_H__
+#define __GST_FDKAACDEC_H__
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudiodecoder.h>
+
+#include <fdk-aac/aacdecoder_lib.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_FDKAACDEC \
+  (gst_fdkaacdec_get_type())
+#define GST_FDKAACDEC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_FDKAACDEC, GstFdkAacDec))
+#define GST_FDKAACDEC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_FDKAACDEC, GstFdkAacDecClass))
+#define GST_IS_FDKAACDEC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_FDKAACDEC))
+#define GST_IS_FDKAACDEC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_FDKAACDEC))
+
+typedef struct _GstFdkAacDec GstFdkAacDec;
+typedef struct _GstFdkAacDecClass GstFdkAacDecClass;
+
+struct _GstFdkAacDec {
+  GstAudioDecoder element;
+
+  HANDLE_AACDECODER dec;
+  guint8 *decode_buffer;
+  gint decode_buffer_size;
+};
+
+struct _GstFdkAacDecClass {
+  GstAudioDecoderClass parent_class;
+};
+
+GType gst_fdkaacdec_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_FDKAACDEC_H__ */

ext/fdkaac/plugin.c

@@ -23,12 +23,15 @@
 
 #include <gst/gst.h>
 #include "gstfdkaacenc.h"
+#include "gstfdkaacdec.h"
 
 static gboolean
 plugin_init (GstPlugin * plugin)
 {
   return gst_element_register (plugin, "fdkaacenc", GST_RANK_PRIMARY,
-      GST_TYPE_FDKAACENC);
+      GST_TYPE_FDKAACENC)
+      && gst_element_register (plugin, "fdkaacdec", GST_RANK_MARGINAL,
+      GST_TYPE_FDKAACDEC);
 }
 
 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,