6, 0, 0, 0, 0, 1, 1
};
- static gboolean gst_amrwbdec_event (GstPad * pad, GstEvent * event);
- static GstFlowReturn gst_amrwbdec_chain (GstPad * pad, GstBuffer * buffer);
- static gboolean gst_amrwbdec_setcaps (GstPad * pad, GstCaps * caps);
- static GstStateChangeReturn gst_amrwbdec_state_change (GstElement * element,
- GstStateChange transition);
-
- static void gst_amrwbdec_finalize (GObject * object);
+ static gboolean gst_amrwbdec_start (GstAudioDecoder * dec);
+ static gboolean gst_amrwbdec_stop (GstAudioDecoder * dec);
+ static gboolean gst_amrwbdec_set_format (GstAudioDecoder * dec, GstCaps * caps);
+ static gboolean gst_amrwbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
+ gint * offset, gint * length);
+ static GstFlowReturn gst_amrwbdec_handle_frame (GstAudioDecoder * dec,
+ GstBuffer * buffer);
-#define _do_init(bla) \
- GST_DEBUG_CATEGORY_INIT (gst_amrwbdec_debug, "amrwbdec", 0, "AMR-WB audio decoder");
-
-GST_BOILERPLATE_FULL (GstAmrwbDec, gst_amrwbdec, GstAudioDecoder,
- GST_TYPE_AUDIO_DECODER, _do_init);
+#define gst_amrwbdec_parent_class parent_class
- G_DEFINE_TYPE (GstAmrwbDec, gst_amrwbdec, GST_TYPE_ELEMENT);
++G_DEFINE_TYPE (GstAmrwbDec, gst_amrwbdec, GST_TYPE_AUDIO_DECODER);
static void
-gst_amrwbdec_base_init (gpointer klass)
+gst_amrwbdec_class_init (GstAmrwbDecClass * klass)
{
- GObjectClass *object_class = G_OBJECT_CLASS (klass);
++ GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
- object_class->finalize = gst_amrwbdec_finalize;
-
- element_class->change_state = GST_DEBUG_FUNCPTR (gst_amrwbdec_state_change);
-
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_add_pad_template (element_class,
"Codec/Decoder/Audio",
"Adaptive Multi-Rate Wideband audio decoder",
"Renato Araujo <renato.filho@indt.org.br>");
-}
-
-static void
-gst_amrwbdec_class_init (GstAmrwbDecClass * klass)
-{
- GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
+ base_class->start = GST_DEBUG_FUNCPTR (gst_amrwbdec_start);
+ base_class->stop = GST_DEBUG_FUNCPTR (gst_amrwbdec_stop);
+ base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrwbdec_set_format);
+ base_class->parse = GST_DEBUG_FUNCPTR (gst_amrwbdec_parse);
+ base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrwbdec_handle_frame);
++
+ GST_DEBUG_CATEGORY_INIT (gst_amrwbdec_debug, "amrwbdec", 0,
+ "AMR-WB audio decoder");
}
static void
-gst_amrwbdec_init (GstAmrwbDec * amrwbdec, GstAmrwbDecClass * klass)
+gst_amrwbdec_init (GstAmrwbDec * amrwbdec)
{
- /* create the sink pad */
- amrwbdec->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
- gst_pad_set_event_function (amrwbdec->sinkpad, gst_amrwbdec_event);
- gst_pad_set_chain_function (amrwbdec->sinkpad, gst_amrwbdec_chain);
- gst_element_add_pad (GST_ELEMENT (amrwbdec), amrwbdec->sinkpad);
+ }
- /* create the src pad */
- amrwbdec->srcpad = gst_pad_new_from_static_template (&src_template, "src");
- gst_pad_use_fixed_caps (amrwbdec->srcpad);
- gst_element_add_pad (GST_ELEMENT (amrwbdec), amrwbdec->srcpad);
+ static gboolean
+ gst_amrwbdec_start (GstAudioDecoder * dec)
+ {
+ GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
- amrwbdec->adapter = gst_adapter_new ();
+ GST_DEBUG_OBJECT (dec, "start");
+ if (!(amrwbdec->handle = D_IF_init ()))
+ return FALSE;
- /* init rest */
- amrwbdec->handle = NULL;
- amrwbdec->channels = 0;
amrwbdec->rate = 0;
- amrwbdec->duration = 0;
- amrwbdec->ts = -1;
+ amrwbdec->channels = 0;
+
+ return TRUE;
}
- static void
- gst_amrwbdec_finalize (GObject * object)
+ static gboolean
+ gst_amrwbdec_stop (GstAudioDecoder * dec)
{
- GstAmrwbDec *amrwbdec;
+ GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
- amrwbdec = GST_AMRWBDEC (object);
+ GST_DEBUG_OBJECT (dec, "stop");
+ D_IF_exit (amrwbdec->handle);
- gst_adapter_clear (amrwbdec->adapter);
- g_object_unref (amrwbdec->adapter);
-
- G_OBJECT_CLASS (parent_class)->finalize (object);
+ return TRUE;
}
static gboolean
gst_structure_get_int (structure, "rate", &amrwbdec->rate);
/* create reverse caps */
-- copy = gst_caps_new_simple ("audio/x-raw-int",
++ copy = gst_caps_new_simple ("audio/x-raw",
++ "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
"channels", G_TYPE_INT, amrwbdec->channels,
-- "width", G_TYPE_INT, 16,
-- "depth", G_TYPE_INT, 16,
-- "endianness", G_TYPE_INT, G_BYTE_ORDER,
-- "rate", G_TYPE_INT, amrwbdec->rate, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
++ "rate", G_TYPE_INT, amrwbdec->rate, NULL);
- amrwbdec->duration = gst_util_uint64_scale_int (GST_SECOND, L_FRAME16k,
- amrwbdec->rate * amrwbdec->channels);
-
- gst_pad_set_caps (amrwbdec->srcpad, copy);
- gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (amrwbdec), copy);
++ gst_audio_decoder_set_outcaps (dec, copy);
gst_caps_unref (copy);
- gst_object_unref (amrwbdec);
-
return TRUE;
}
- static gboolean
- gst_amrwbdec_event (GstPad * pad, GstEvent * event)
+ static GstFlowReturn
+ gst_amrwbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
+ gint * offset, gint * length)
{
- GstAmrwbDec *amrwbdec;
- gboolean ret = TRUE;
-
- amrwbdec = GST_AMRWBDEC (gst_pad_get_parent (pad));
-
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_CAPS:
- {
- GstCaps *caps;
-
- gst_event_parse_caps (event, &caps);
- ret = gst_amrwbdec_setcaps (pad, caps);
- gst_event_unref (event);
- break;
- }
- case GST_EVENT_FLUSH_START:
- ret = gst_pad_push_event (amrwbdec->srcpad, event);
- break;
- case GST_EVENT_FLUSH_STOP:
- ret = gst_pad_push_event (amrwbdec->srcpad, event);
- gst_adapter_clear (amrwbdec->adapter);
- amrwbdec->ts = -1;
- break;
- case GST_EVENT_EOS:
- gst_adapter_clear (amrwbdec->adapter);
- ret = gst_pad_push_event (amrwbdec->srcpad, event);
- break;
- case GST_EVENT_SEGMENT:
- {
- GstSegment seg;
-
- gst_event_copy_segment (event, &seg);
- /* we need time for now */
- if (seg.format != GST_FORMAT_TIME)
- goto newseg_wrong_format;
-
- GST_DEBUG_OBJECT (amrwbdec, "segment: %" GST_SEGMENT_FORMAT, &seg);
-
- /* now configure the values */
- amrwbdec->segment = seg;
-
- ret = gst_pad_push_event (amrwbdec->srcpad, event);
- }
- break;
- default:
- ret = gst_pad_push_event (amrwbdec->srcpad, event);
- break;
+ GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
- const guint8 *data;
++ guint8 header[1];
+ guint size;
+ gboolean sync, eos;
+ gint block, mode;
+
+ size = gst_adapter_available (adapter);
+ g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
+
+ gst_audio_decoder_get_parse_state (dec, &sync, &eos);
+
+ /* need to peek data to get the size */
+ if (gst_adapter_available (adapter) < 1)
+ return GST_FLOW_ERROR;
+
- data = gst_adapter_peek (adapter, 1);
- mode = (data[0] >> 3) & 0x0F;
++ gst_adapter_copy (adapter, header, 0, 1);
++ mode = (header[0] >> 3) & 0x0F;
+ block = block_size[mode];
+
+ GST_DEBUG_OBJECT (amrwbdec, "mode %d, block %d", mode, block);
+
+ if (block) {
+ *offset = 0;
+ *length = block;
+ } else {
+ /* no frame yet, skip one byte */
+ GST_LOG_OBJECT (amrwbdec, "skipping byte");
+ *offset = 1;
+ return GST_FLOW_UNEXPECTED;
}
- done:
- gst_object_unref (amrwbdec);
- return ret;
-
- /* ERRORS */
- newseg_wrong_format:
- {
- GST_DEBUG_OBJECT (amrwbdec, "received non TIME newsegment");
- goto done;
- }
+ return GST_FLOW_OK;
}
static GstFlowReturn
- gst_amrwbdec_chain (GstPad * pad, GstBuffer * buffer)
+ gst_amrwbdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
{
GstAmrwbDec *amrwbdec;
- GstFlowReturn ret = GST_FLOW_OK;
+ GstBuffer *out;
- const guint8 *data;
++ guint8 *data;
++ Word16 *outdata;
+
+ amrwbdec = GST_AMRWBDEC (dec);
- amrwbdec = GST_AMRWBDEC (gst_pad_get_parent (pad));
+ /* no fancy flushing */
- if (!buffer || !GST_BUFFER_SIZE (buffer))
++ if (!buffer || !gst_buffer_get_size (buffer))
+ return GST_FLOW_OK;
if (amrwbdec->rate == 0 || amrwbdec->channels == 0)
goto not_negotiated;
- /* discontinuity, don't combine samples before and after the
- * DISCONT */
- if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
- gst_adapter_clear (amrwbdec->adapter);
- amrwbdec->ts = -1;
- amrwbdec->discont = TRUE;
- }
+ /* the library seems to write into the source data, hence the copy. */
+ /* should be no problem */
- data = GST_BUFFER_DATA (buffer);
++ data = gst_buffer_map (buffer, NULL, NULL, GST_MAP_READ);
- /* take latest timestamp, FIXME timestamp is the one of the
- * first buffer in the adapter. */
- if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
- amrwbdec->ts = GST_BUFFER_TIMESTAMP (buffer);
+ /* get output */
+ out = gst_buffer_new_and_alloc (sizeof (gint16) * L_FRAME16k);
++ outdata = gst_buffer_map (out, NULL, NULL, GST_MAP_WRITE);
- gst_adapter_push (amrwbdec->adapter, buffer);
+ /* decode */
- D_IF_decode (amrwbdec->handle, (unsigned char *) data,
- (Word16 *) GST_BUFFER_DATA (out), _good_frame);
++ D_IF_decode (amrwbdec->handle, (unsigned char *) data, outdata, _good_frame);
+
- while (TRUE) {
- GstBuffer *out;
- guint8 head[1];
- const guint8 *data;
- Word16 *out_data;
- gint block, mode;
++ gst_buffer_unmap (out, outdata, sizeof (gint16) * L_FRAME16k);
++ gst_buffer_unmap (buffer, data, -1);
- /* need to peek data to get the size */
- if (gst_adapter_available (amrwbdec->adapter) < 1)
- break;
- gst_adapter_copy (amrwbdec->adapter, head, 0, 1);
-
- /* get size */
- mode = (head[0] >> 3) & 0x0F;
- block = block_size[mode];
-
- GST_DEBUG_OBJECT (amrwbdec, "mode %d, block %d", mode, block);
-
- if (!block) {
- GST_LOG_OBJECT (amrwbdec, "skipping byte");
- gst_adapter_flush (amrwbdec->adapter, 1);
- continue;
- }
-
- if (gst_adapter_available (amrwbdec->adapter) < block)
- break;
-
- /* the library seems to write into the source data, hence the copy. */
- data = gst_adapter_take (amrwbdec->adapter, block);
-
- /* get output */
- out = gst_buffer_new_and_alloc (sizeof (gint16) * L_FRAME16k);
-
- GST_BUFFER_DURATION (out) = amrwbdec->duration;
- GST_BUFFER_TIMESTAMP (out) = amrwbdec->ts;
-
- if (amrwbdec->ts != -1)
- amrwbdec->ts += amrwbdec->duration;
- if (amrwbdec->discont) {
- GST_BUFFER_FLAG_SET (out, GST_BUFFER_FLAG_DISCONT);
- amrwbdec->discont = FALSE;
- }
-
- /* decode */
- out_data = gst_buffer_map (out, NULL, NULL, GST_MAP_WRITE);
- D_IF_decode (amrwbdec->handle, (unsigned char *) data,
- out_data, _good_frame);
- gst_buffer_unmap (out, out_data, -1);
-
- g_free ((gpointer) data);
-
- /* send out */
- ret = gst_pad_push (amrwbdec->srcpad, out);
- }
-
- gst_object_unref (amrwbdec);
- return ret;
+ /* send out */
+ return gst_audio_decoder_finish_frame (dec, out, 1);
/* ERRORS */
not_negotiated:
projects / platform / upstream / gstreamer.git / commitdiff