--- /dev/null
+/*
+ * Copyright (C) 2014, Sebastian Dröge <sebastian@centricular.com>
+ *
+ * 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
+ *
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+
+#include "gstomxaacdec.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_omx_aac_dec_debug_category);
+#define GST_CAT_DEFAULT gst_omx_aac_dec_debug_category
+
+/* prototypes */
+static gboolean gst_omx_aac_dec_set_format (GstOMXAudioDec * dec,
+ GstOMXPort * port, GstCaps * caps);
+static gboolean gst_omx_aac_dec_is_format_change (GstOMXAudioDec * dec,
+ GstOMXPort * port, GstCaps * caps);
+static gint gst_omx_aac_dec_get_samples_per_frame (GstOMXAudioDec * dec,
+ GstOMXPort * port);
+
+/* class initialization */
+
+#define DEBUG_INIT \
+ GST_DEBUG_CATEGORY_INIT (gst_omx_aac_dec_debug_category, "omxaacdec", 0, \
+ "debug category for gst-omx aac audio decoder");
+
+G_DEFINE_TYPE_WITH_CODE (GstOMXAACDec, gst_omx_aac_dec,
+ GST_TYPE_OMX_AUDIO_DEC, DEBUG_INIT);
+
+static void
+gst_omx_aac_dec_class_init (GstOMXAACDecClass * klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GstOMXAudioDecClass *audiodec_class = GST_OMX_AUDIO_DEC_CLASS (klass);
+
+ audiodec_class->set_format = GST_DEBUG_FUNCPTR (gst_omx_aac_dec_set_format);
+ audiodec_class->is_format_change =
+ GST_DEBUG_FUNCPTR (gst_omx_aac_dec_is_format_change);
+ audiodec_class->get_samples_per_frame =
+ GST_DEBUG_FUNCPTR (gst_omx_aac_dec_get_samples_per_frame);
+
+ audiodec_class->cdata.default_sink_template_caps = "audio/mpeg, "
+ "mpegversion=(int){2, 4}, "
+ "stream-format=(string) { raw, adts, adif, loas }, "
+ "rate=(int)[8000,48000], "
+ "channels=(int)[1,9], " "framed=(boolean) true";
+
+ gst_element_class_set_static_metadata (element_class,
+ "OpenMAX AAC Audio Decoder",
+ "Codec/Decoder/Audio",
+ "Decode AAC audio streams",
+ "Sebastian Dröge <sebastian@centricular.com>");
+
+ gst_omx_set_default_role (&audiodec_class->cdata, "audio_decoder.aac");
+}
+
+static void
+gst_omx_aac_dec_init (GstOMXAACDec * self)
+{
+ /* FIXME: Other values exist too! */
+ self->spf = 1024;
+}
+
+static gboolean
+gst_omx_aac_dec_set_format (GstOMXAudioDec * dec, GstOMXPort * port,
+ GstCaps * caps)
+{
+ GstOMXAACDec *self = GST_OMX_AAC_DEC (dec);
+ OMX_PARAM_PORTDEFINITIONTYPE port_def;
+ OMX_AUDIO_PARAM_AACPROFILETYPE aac_param;
+ OMX_ERRORTYPE err;
+ GstStructure *s;
+ gint rate, channels, mpegversion;
+ const gchar *stream_format;
+
+ gst_omx_port_get_port_definition (port, &port_def);
+ port_def.format.audio.eEncoding = OMX_AUDIO_CodingAAC;
+ err = gst_omx_port_update_port_definition (port, &port_def);
+ if (err != OMX_ErrorNone) {
+ GST_ERROR_OBJECT (self,
+ "Failed to set AAC format on component: %s (0x%08x)",
+ gst_omx_error_to_string (err), err);
+ return FALSE;
+ }
+
+ GST_OMX_INIT_STRUCT (&aac_param);
+ aac_param.nPortIndex = port->index;
+
+ err =
+ gst_omx_component_get_parameter (dec->dec, OMX_IndexParamAudioAac,
+ &aac_param);
+ if (err != OMX_ErrorNone) {
+ GST_ERROR_OBJECT (self,
+ "Failed to get AAC parameters from component: %s (0x%08x)",
+ gst_omx_error_to_string (err), err);
+ return FALSE;
+ }
+
+ s = gst_caps_get_structure (caps, 0);
+
+ if (!gst_structure_get_int (s, "mpegversion", &mpegversion) ||
+ !gst_structure_get_int (s, "rate", &rate) ||
+ !gst_structure_get_int (s, "channels", &channels)) {
+ GST_ERROR_OBJECT (self, "Incomplete caps");
+ return FALSE;
+ }
+
+ stream_format = gst_structure_get_string (s, "stream-format");
+ if (!stream_format) {
+ GST_ERROR_OBJECT (self, "Incomplete caps");
+ return FALSE;
+ }
+
+ aac_param.nChannels = channels;
+ aac_param.nSampleRate = rate;
+ aac_param.nBitRate = 0; /* unknown */
+ aac_param.nAudioBandWidth = 0; /* decoder decision */
+ aac_param.eChannelMode = 0; /* FIXME */
+ if (mpegversion == 2)
+ aac_param.eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP2ADTS;
+ else if (strcmp (stream_format, "adts") == 0)
+ aac_param.eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP4ADTS;
+ else if (strcmp (stream_format, "loas") == 0)
+ aac_param.eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP4LOAS;
+ else if (strcmp (stream_format, "adif") == 0)
+ aac_param.eAACStreamFormat = OMX_AUDIO_AACStreamFormatADIF;
+ else if (strcmp (stream_format, "raw") == 0)
+ aac_param.eAACStreamFormat = OMX_AUDIO_AACStreamFormatRAW;
+ else
+ aac_param.eAACStreamFormat = OMX_AUDIO_AACStreamFormatRAW;
+
+ err =
+ gst_omx_component_set_parameter (dec->dec, OMX_IndexParamAudioAac,
+ &aac_param);
+ if (err != OMX_ErrorNone) {
+ GST_ERROR_OBJECT (self, "Error setting AAC parameters: %s (0x%08x)",
+ gst_omx_error_to_string (err), err);
+ return FALSE;
+ }
+
+ return TRUE;
+}
+
+static gboolean
+gst_omx_aac_dec_is_format_change (GstOMXAudioDec * dec, GstOMXPort * port,
+ GstCaps * caps)
+{
+ GstOMXAACDec *self = GST_OMX_AAC_DEC (dec);
+ OMX_AUDIO_PARAM_AACPROFILETYPE aac_param;
+ OMX_ERRORTYPE err;
+ GstStructure *s;
+ gint rate, channels, mpegversion;
+ const gchar *stream_format;
+
+ GST_OMX_INIT_STRUCT (&aac_param);
+ aac_param.nPortIndex = port->index;
+
+ err =
+ gst_omx_component_get_parameter (dec->dec, OMX_IndexParamAudioAac,
+ &aac_param);
+ if (err != OMX_ErrorNone) {
+ GST_ERROR_OBJECT (self,
+ "Failed to get AAC parameters from component: %s (0x%08x)",
+ gst_omx_error_to_string (err), err);
+ return FALSE;
+ }
+
+ s = gst_caps_get_structure (caps, 0);
+
+ if (!gst_structure_get_int (s, "mpegversion", &mpegversion) ||
+ !gst_structure_get_int (s, "rate", &rate) ||
+ !gst_structure_get_int (s, "channels", &channels)) {
+ GST_ERROR_OBJECT (self, "Incomplete caps");
+ return FALSE;
+ }
+
+ stream_format = gst_structure_get_string (s, "stream-format");
+ if (!stream_format) {
+ GST_ERROR_OBJECT (self, "Incomplete caps");
+ return FALSE;
+ }
+
+ if (aac_param.nChannels != channels)
+ return TRUE;
+
+ if (aac_param.nSampleRate != rate)
+ return TRUE;
+
+ if (mpegversion == 2
+ && aac_param.eAACStreamFormat != OMX_AUDIO_AACStreamFormatMP2ADTS)
+ return TRUE;
+ if (aac_param.eAACStreamFormat == OMX_AUDIO_AACStreamFormatMP4ADTS &&
+ strcmp (stream_format, "adts") != 0)
+ return TRUE;
+ if (aac_param.eAACStreamFormat == OMX_AUDIO_AACStreamFormatMP4LOAS &&
+ strcmp (stream_format, "loas") != 0)
+ return TRUE;
+ if (aac_param.eAACStreamFormat == OMX_AUDIO_AACStreamFormatADIF &&
+ strcmp (stream_format, "adif") != 0)
+ return TRUE;
+ if (aac_param.eAACStreamFormat == OMX_AUDIO_AACStreamFormatRAW &&
+ strcmp (stream_format, "raw") != 0)
+ return TRUE;
+
+ return FALSE;
+}
+
+static gint
+gst_omx_aac_dec_get_samples_per_frame (GstOMXAudioDec * dec, GstOMXPort * port)
+{
+ return GST_OMX_AAC_DEC (dec)->spf;
+}
--- /dev/null
+/*
+ * Copyright (C) 2014, Sebastian Dröge <sebastian@centricular.com>
+ *
+ * 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_OMX_AAC_DEC_H__
+#define __GST_OMX_AAC_DEC_H__
+
+#include <gst/gst.h>
+#include "gstomxaudiodec.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OMX_AAC_DEC \
+ (gst_omx_aac_dec_get_type())
+#define GST_OMX_AAC_DEC(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OMX_AAC_DEC,GstOMXAACDec))
+#define GST_OMX_AAC_DEC_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OMX_AAC_DEC,GstOMXAACDecClass))
+#define GST_OMX_AAC_DEC_GET_CLASS(obj) \
+ (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_OMX_AAC_DEC,GstOMXAACDecClass))
+#define GST_IS_OMX_AAC_DEC(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OMX_AAC_DEC))
+#define GST_IS_OMX_AAC_DEC_CLASS(obj) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OMX_AAC_DEC))
+
+typedef struct _GstOMXAACDec GstOMXAACDec;
+typedef struct _GstOMXAACDecClass GstOMXAACDecClass;
+
+struct _GstOMXAACDec
+{
+ GstOMXAudioDec parent;
+ gint spf;
+};
+
+struct _GstOMXAACDecClass
+{
+ GstOMXAudioDecClass parent_class;
+};
+
+GType gst_omx_aac_dec_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_OMX_AAC_DEC_H__ */
+