Add support for audio decoders, completely untested so far

sys/androidmedia/gstamc-constants.h

@@ -201,4 +201,27 @@ enum
   AACObjectELD = 39
 };
 
+/* Copies from AudioFormat.java */
+enum
+{
+  CHANNEL_OUT_FRONT_LEFT = 0x4,
+  CHANNEL_OUT_FRONT_RIGHT = 0x8,
+  CHANNEL_OUT_FRONT_CENTER = 0x10,
+  CHANNEL_OUT_LOW_FREQUENCY = 0x20,
+  CHANNEL_OUT_BACK_LEFT = 0x40,
+  CHANNEL_OUT_BACK_RIGHT = 0x80,
+  CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x100,
+  CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x200,
+  CHANNEL_OUT_BACK_CENTER = 0x400,
+  CHANNEL_OUT_SIDE_LEFT = 0x800,
+  CHANNEL_OUT_SIDE_RIGHT = 0x1000,
+  CHANNEL_OUT_TOP_CENTER = 0x2000,
+  CHANNEL_OUT_TOP_FRONT_LEFT = 0x4000,
+  CHANNEL_OUT_TOP_FRONT_CENTER = 0x8000,
+  CHANNEL_OUT_TOP_FRONT_RIGHT = 0x10000,
+  CHANNEL_OUT_TOP_BACK_LEFT = 0x20000,
+  CHANNEL_OUT_TOP_BACK_CENTER = 0x40000,
+  CHANNEL_OUT_TOP_BACK_RIGHT = 0x80000
+};
+
 #endif

sys/androidmedia/gstamc.c

@@ -26,9 +26,11 @@
 #include "gstamc-constants.h"
 
 #include "gstamcvideodec.h"
+#include "gstamcaudiodec.h"
 
 #include <gst/gst.h>
 #include <gst/video/video.h>
+#include <gst/audio/audio.h>
 #include <string.h>
 #include <jni.h>
 
@@ -2468,6 +2470,107 @@ gst_amc_aac_profile_from_string (const gchar * profile)
   return -1;
 }
 
+static const struct
+{
+  guint32 mask;
+  GstAudioChannelPosition pos;
+} channel_mapping_table[] = {
+  {
+  CHANNEL_OUT_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT}, {
+  CHANNEL_OUT_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, {
+  CHANNEL_OUT_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, {
+  CHANNEL_OUT_LOW_FREQUENCY, GST_AUDIO_CHANNEL_POSITION_LFE}, {
+  CHANNEL_OUT_BACK_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT}, {
+  CHANNEL_OUT_BACK_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
+  CHANNEL_OUT_FRONT_LEFT_OF_CENTER,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER}, {
+  CHANNEL_OUT_FRONT_RIGHT_OF_CENTER,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER}, {
+  CHANNEL_OUT_BACK_CENTER, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, {
+  CHANNEL_OUT_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT}, {
+  CHANNEL_OUT_SIDE_RIGHT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}, {
+  CHANNEL_OUT_TOP_CENTER, GST_AUDIO_CHANNEL_POSITION_INVALID}, {
+  CHANNEL_OUT_TOP_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_INVALID}, {
+  CHANNEL_OUT_TOP_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_INVALID}, {
+  CHANNEL_OUT_TOP_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_INVALID}, {
+  CHANNEL_OUT_TOP_BACK_LEFT, GST_AUDIO_CHANNEL_POSITION_INVALID}, {
+  CHANNEL_OUT_TOP_BACK_CENTER, GST_AUDIO_CHANNEL_POSITION_INVALID}, {
+  CHANNEL_OUT_TOP_BACK_RIGHT, GST_AUDIO_CHANNEL_POSITION_INVALID}
+};
+
+GstAudioChannelPosition *
+gst_amc_audio_channel_mask_to_positions (guint32 channel_mask, gint channels)
+{
+  GstAudioChannelPosition *pos = g_new0 (GstAudioChannelPosition, channels);
+  gint i, j;
+
+  if (channel_mask == 0 && channels == 1) {
+    pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
+    return pos;
+  }
+
+  if (channel_mask == 0 && channels == 2) {
+    pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+    pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+    return pos;
+  }
+
+  for (i = 0, j = 0; i < G_N_ELEMENTS (channel_mapping_table); i++) {
+    if ((channel_mask & channel_mapping_table[i].mask)) {
+      pos[j++] = channel_mapping_table[i].pos;
+      if (channel_mapping_table[i].pos == GST_AUDIO_CHANNEL_POSITION_INVALID) {
+        g_free (pos);
+        GST_ERROR ("Unable to map channel mask 0x%08x",
+            channel_mapping_table[i].mask);
+        return NULL;
+      }
+      if (j == channels)
+        break;
+    }
+  }
+
+  if (j != channels) {
+    g_free (pos);
+    GST_ERROR ("Unable to map all channel positions in mask 0x%08x",
+        channel_mask);
+    return NULL;
+  }
+
+  return pos;
+}
+
+guint32
+gst_amc_audio_channel_mask_from_positions (GstAudioChannelPosition * positions,
+    gint channels)
+{
+  gint i, j;
+  guint32 channel_mask = 0;
+
+  if (channels == 1 && !positions)
+    return CHANNEL_OUT_FRONT_CENTER;
+  if (channels == 2 && !positions)
+    return CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT;
+
+  for (i = 0; i < channels; i++) {
+    if (positions[i] == GST_AUDIO_CHANNEL_POSITION_INVALID)
+      return 0;
+
+    for (j = 0; j < G_N_ELEMENTS (channel_mapping_table); j++) {
+      if (channel_mapping_table[j].pos == positions[i]) {
+        channel_mask |= channel_mapping_table[j].mask;
+        break;
+      }
+    }
+
+    if (j == G_N_ELEMENTS (channel_mapping_table)) {
+      GST_ERROR ("Unable to map channel position %d", positions[i]);
+      return 0;
+    }
+  }
+
+  return channel_mask;
+}
+
 static gchar *
 create_type_name (const gchar * parent_name, const gchar * codec_name)
 {
@@ -2579,6 +2682,8 @@ register_codecs (GstPlugin * plugin)
 
       if (is_video && !codec_info->is_encoder) {
         type = gst_amc_video_dec_get_type ();
+      } else if (is_audio && !codec_info->is_encoder) {
+        type = gst_amc_audio_dec_get_type ();
       } else {
         GST_DEBUG ("Skipping unsupported codec type");
         continue;

sys/androidmedia/gstamc.h

@@ -23,6 +23,7 @@
 
 #include <gst/gst.h>
 #include <gst/video/video.h>
+#include <gst/audio/multichannel.h>
 #include <jni.h>
 
 G_BEGIN_DECLS
@@ -136,6 +137,9 @@ gint gst_amc_mpeg4_level_from_string (const gchar *level);
 const gchar * gst_amc_aac_profile_to_string (gint profile);
 gint gst_amc_aac_profile_from_string (const gchar *profile);
 
+GstAudioChannelPosition* gst_amc_audio_channel_mask_to_positions (guint32 channel_mask, gint channels);
+guint32 gst_amc_audio_channel_mask_from_positions (GstAudioChannelPosition *positions, gint channels);
+
 G_END_DECLS
 
 #endif /* __GST_AMC_H__ */

sys/androidmedia/gstamcaudiodec.h

@@ -0,0 +1,96 @@
+/*
+ * Copyright (C) 2012, Collabora Ltd.
+ *   Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation
+ * version 2.1 of the License.
+ *
+ * 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301 USA
+ *
+ */
+
+#ifndef __GST_AMC_AUDIO_DEC_H__
+#define __GST_AMC_AUDIO_DEC_H__
+
+#include <gst/gst.h>
+#include <gst/audio/multichannel.h>
+#include <gst/audio/gstaudiodecoder.h>
+
+#include "gstamc.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_AMC_AUDIO_DEC \
+  (gst_amc_audio_dec_get_type())
+#define GST_AMC_AUDIO_DEC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AMC_AUDIO_DEC,GstAmcAudioDec))
+#define GST_AMC_AUDIO_DEC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AMC_AUDIO_DEC,GstAmcAudioDecClass))
+#define GST_AMC_AUDIO_DEC_GET_CLASS(obj) \
+  (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_AMC_AUDIO_DEC,GstAmcAudioDecClass))
+#define GST_IS_AMC_AUDIO_DEC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AMC_AUDIO_DEC))
+#define GST_IS_AMC_AUDIO_DEC_CLASS(obj) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AMC_AUDIO_DEC))
+
+typedef struct _GstAmcAudioDec GstAmcAudioDec;
+typedef struct _GstAmcAudioDecClass GstAmcAudioDecClass;
+
+struct _GstAmcAudioDec
+{
+  GstAudioDecoder parent;
+
+  /* < private > */
+  GstAmcCodec *codec;
+  GstAmcBuffer *input_buffers, *output_buffers;
+  gsize n_input_buffers, n_output_buffers;
+
+  GstCaps *input_caps;
+  GList *codec_datas;
+  gboolean input_caps_changed;
+
+  /* Output format of the codec */
+  gint channels, rate;
+  GstAudioChannelPosition *positions;
+
+  GstBuffer *codec_data;
+  /* TRUE if the component is configured and saw
+   * the first buffer */
+  gboolean started;
+  gboolean flushing;
+
+  GstClockTime last_upstream_ts;
+
+  /* Draining state */
+  GMutex *drain_lock;
+  GCond *drain_cond;
+  /* TRUE if EOS buffers shouldn't be forwarded */
+  gboolean draining;
+
+  /* TRUE if upstream is EOS */
+  gboolean eos;
+
+  GstFlowReturn downstream_flow_ret;
+};
+
+struct _GstAmcAudioDecClass
+{
+  GstAudioDecoderClass parent_class;
+
+  const GstAmcCodecInfo *codec_info;
+};
+
+GType gst_amc_audio_dec_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_AMC_AUDIO_DEC_H__ */