mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-24 17:20:36 +00:00
Merge branch 'master' into 0.11
Conflicts: ext/amrnb/amrnbdec.c
This commit is contained in:
commit
76a13cbcdc
2 changed files with 100 additions and 248 deletions
|
@ -96,34 +96,26 @@ static void gst_amrnbdec_set_property (GObject * object, guint prop_id,
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static void gst_amrnbdec_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_amrnbdec_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_amrnbdec_chain (GstPad * pad, GstBuffer * buffer);
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static gboolean gst_amrnbdec_setcaps (GstPad * pad, GstCaps * caps);
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static GstStateChangeReturn gst_amrnbdec_state_change (GstElement * element,
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GstStateChange transition);
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static void gst_amrnbdec_finalize (GObject * object);
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static gboolean gst_amrnbdec_start (GstAudioDecoder * dec);
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static gboolean gst_amrnbdec_stop (GstAudioDecoder * dec);
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static gboolean gst_amrnbdec_set_format (GstAudioDecoder * dec, GstCaps * caps);
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static gboolean gst_amrnbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
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gint * offset, gint * length);
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static GstFlowReturn gst_amrnbdec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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#define gst_amrnbdec_parent_class parent_class
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G_DEFINE_TYPE (GstAmrnbDec, gst_amrnbdec, GST_TYPE_ELEMENT);
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G_DEFINE_TYPE (GstAmrnbDec, gst_amrnbdec, GST_TYPE_AUDIO_DECODER);
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static void
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gst_amrnbdec_class_init (GstAmrnbDecClass * klass)
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{
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
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object_class->set_property = gst_amrnbdec_set_property;
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object_class->get_property = gst_amrnbdec_get_property;
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object_class->finalize = gst_amrnbdec_finalize;
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g_object_class_install_property (object_class, PROP_VARIANT,
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g_param_spec_enum ("variant", "Variant",
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"The decoder variant", GST_AMRNB_VARIANT_TYPE,
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VARIANT_DEFAULT,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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element_class->change_state = GST_DEBUG_FUNCPTR (gst_amrnbdec_state_change);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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@ -135,6 +127,18 @@ gst_amrnbdec_class_init (GstAmrnbDecClass * klass)
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"Adaptive Multi-Rate Narrow-Band audio decoder",
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"GStreamer maintainers <gstreamer-devel@lists.sourceforge.net>");
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base_class->start = GST_DEBUG_FUNCPTR (gst_amrnbdec_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_amrnbdec_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrnbdec_set_format);
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base_class->parse = GST_DEBUG_FUNCPTR (gst_amrnbdec_parse);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrnbdec_handle_frame);
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g_object_class_install_property (object_class, PROP_VARIANT,
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g_param_spec_enum ("variant", "Variant",
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"The decoder variant", GST_AMRNB_VARIANT_TYPE,
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VARIANT_DEFAULT,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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GST_DEBUG_CATEGORY_INIT (gst_amrnbdec_debug, "amrnbdec", 0,
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"AMR-NB audio decoder");
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}
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@ -142,34 +146,32 @@ gst_amrnbdec_class_init (GstAmrnbDecClass * klass)
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static void
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gst_amrnbdec_init (GstAmrnbDec * amrnbdec)
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{
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/* create the sink pad */
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amrnbdec->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
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gst_pad_set_event_function (amrnbdec->sinkpad, gst_amrnbdec_event);
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gst_pad_set_chain_function (amrnbdec->sinkpad, gst_amrnbdec_chain);
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gst_element_add_pad (GST_ELEMENT (amrnbdec), amrnbdec->sinkpad);
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/* create the src pad */
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amrnbdec->srcpad = gst_pad_new_from_static_template (&src_template, "src");
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gst_pad_use_fixed_caps (amrnbdec->srcpad);
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gst_element_add_pad (GST_ELEMENT (amrnbdec), amrnbdec->srcpad);
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amrnbdec->adapter = gst_adapter_new ();
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/* init rest */
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amrnbdec->handle = NULL;
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}
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static void
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gst_amrnbdec_finalize (GObject * object)
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static gboolean
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gst_amrnbdec_start (GstAudioDecoder * dec)
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{
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GstAmrnbDec *amrnbdec;
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GstAmrnbDec *amrnbdec = GST_AMRNBDEC (dec);
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amrnbdec = GST_AMRNBDEC (object);
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GST_DEBUG_OBJECT (dec, "start");
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if (!(amrnbdec->handle = Decoder_Interface_init ()))
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return FALSE;
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gst_adapter_clear (amrnbdec->adapter);
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g_object_unref (amrnbdec->adapter);
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amrnbdec->rate = 0;
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amrnbdec->channels = 0;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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return TRUE;
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}
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static gboolean
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gst_amrnbdec_stop (GstAudioDecoder * dec)
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{
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GstAmrnbDec *amrnbdec = GST_AMRNBDEC (dec);
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GST_DEBUG_OBJECT (dec, "stop");
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Decoder_Interface_exit (amrnbdec->handle);
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return TRUE;
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}
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static void
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@ -207,13 +209,13 @@ gst_amrnbdec_get_property (GObject * object, guint prop_id,
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}
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static gboolean
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gst_amrnbdec_setcaps (GstPad * pad, GstCaps * caps)
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gst_amrnbdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
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{
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GstStructure *structure;
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GstAmrnbDec *amrnbdec;
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GstCaps *copy;
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amrnbdec = GST_AMRNBDEC (gst_pad_get_parent (pad));
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amrnbdec = GST_AMRNBDEC (dec);
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structure = gst_caps_get_structure (caps, 0);
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@ -222,128 +224,36 @@ gst_amrnbdec_setcaps (GstPad * pad, GstCaps * caps)
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gst_structure_get_int (structure, "rate", &amrnbdec->rate);
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/* create reverse caps */
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copy = gst_caps_new_simple ("audio/x-raw-int",
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copy = gst_caps_new_simple ("audio/x-raw",
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"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
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"channels", G_TYPE_INT, amrnbdec->channels,
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"width", G_TYPE_INT, 16,
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"depth", G_TYPE_INT, 16,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"rate", G_TYPE_INT, amrnbdec->rate, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
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amrnbdec->duration = gst_util_uint64_scale_int (GST_SECOND, 160,
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amrnbdec->rate * amrnbdec->channels);
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gst_pad_set_caps (amrnbdec->srcpad, copy);
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"rate", G_TYPE_INT, amrnbdec->rate, NULL);
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gst_audio_decoder_set_outcaps (dec, copy);
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gst_caps_unref (copy);
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gst_object_unref (amrnbdec);
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return TRUE;
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}
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static gboolean
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gst_amrnbdec_event (GstPad * pad, GstEvent * event)
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{
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GstAmrnbDec *amrnbdec;
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gboolean ret = TRUE;
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amrnbdec = GST_AMRNBDEC (gst_pad_get_parent (pad));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_CAPS:
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{
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GstCaps *caps;
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gst_event_parse_caps (event, &caps);
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ret = gst_amrnbdec_setcaps (pad, caps);
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gst_event_unref (event);
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break;
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}
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case GST_EVENT_FLUSH_START:
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ret = gst_pad_push_event (amrnbdec->srcpad, event);
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break;
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case GST_EVENT_FLUSH_STOP:
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ret = gst_pad_push_event (amrnbdec->srcpad, event);
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gst_adapter_clear (amrnbdec->adapter);
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amrnbdec->ts = -1;
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break;
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case GST_EVENT_EOS:
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gst_adapter_clear (amrnbdec->adapter);
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ret = gst_pad_push_event (amrnbdec->srcpad, event);
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break;
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case GST_EVENT_SEGMENT:
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{
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GstSegment seg;
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gst_event_copy_segment (event, &seg);
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/* we need time for now */
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if (seg.format != GST_FORMAT_TIME)
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goto newseg_wrong_format;
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GST_DEBUG_OBJECT (amrnbdec, "segment: %" GST_SEGMENT_FORMAT, &seg);
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/* now configure the values */
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amrnbdec->segment = seg;
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ret = gst_pad_push_event (amrnbdec->srcpad, event);
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break;
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}
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default:
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ret = gst_pad_push_event (amrnbdec->srcpad, event);
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break;
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}
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done:
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gst_object_unref (amrnbdec);
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return ret;
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/* ERRORS */
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newseg_wrong_format:
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{
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GST_DEBUG_OBJECT (amrnbdec, "received non TIME newsegment");
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goto done;
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}
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}
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static GstFlowReturn
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gst_amrnbdec_chain (GstPad * pad, GstBuffer * buffer)
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gst_amrnbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
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gint * offset, gint * length)
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{
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GstAmrnbDec *amrnbdec;
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GstFlowReturn ret;
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amrnbdec = GST_AMRNBDEC (gst_pad_get_parent (pad));
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if (amrnbdec->rate == 0 || amrnbdec->channels == 0)
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goto not_negotiated;
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/* discontinuity, don't combine samples before and after the
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* DISCONT */
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if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
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gst_adapter_clear (amrnbdec->adapter);
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amrnbdec->ts = -1;
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amrnbdec->discont = TRUE;
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}
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/* take latest timestamp, FIXME timestamp is the one of the
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* first buffer in the adapter. */
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if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
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amrnbdec->ts = GST_BUFFER_TIMESTAMP (buffer);
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gst_adapter_push (amrnbdec->adapter, buffer);
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ret = GST_FLOW_OK;
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while (TRUE) {
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GstBuffer *out;
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GstAmrnbDec *amrnbdec = GST_AMRNBDEC (dec);
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guint8 head[1];
|
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guint8 *data;
|
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short *out_data;
|
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guint size;
|
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gboolean sync, eos;
|
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gint block, mode;
|
||||
|
||||
size = gst_adapter_available (adapter);
|
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g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
|
||||
|
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gst_audio_decoder_get_parse_state (dec, &sync, &eos);
|
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|
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/* need to peek data to get the size */
|
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if (gst_adapter_available (amrnbdec->adapter) < 1)
|
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break;
|
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gst_adapter_copy (amrnbdec->adapter, head, 0, 1);
|
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if (gst_adapter_available (adapter) < 1)
|
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return GST_FLOW_ERROR;
|
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|
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gst_adapter_copy (adapter, head, 0, 1);
|
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|
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/* get size */
|
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switch (amrnbdec->variant) {
|
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|
@ -356,43 +266,48 @@ gst_amrnbdec_chain (GstPad * pad, GstBuffer * buffer)
|
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block = block_size_if2[mode] + 1;
|
||||
break;
|
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default:
|
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goto invalid_variant;
|
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g_assert_not_reached ();
|
||||
return GST_FLOW_ERROR;
|
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break;
|
||||
}
|
||||
|
||||
GST_DEBUG_OBJECT (amrnbdec, "mode %d, block %d", mode, block);
|
||||
|
||||
if (!block || gst_adapter_available (amrnbdec->adapter) < block)
|
||||
break;
|
||||
*offset = 0;
|
||||
*length = block;
|
||||
|
||||
/* the library seems to write into the source data, hence
|
||||
* the copy. */
|
||||
data = gst_adapter_take (amrnbdec->adapter, block);
|
||||
return GST_FLOW_OK;
|
||||
}
|
||||
|
||||
static GstFlowReturn
|
||||
gst_amrnbdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
|
||||
{
|
||||
GstAmrnbDec *amrnbdec;
|
||||
guint8 *data;
|
||||
short *out_data;
|
||||
GstBuffer *out;
|
||||
|
||||
amrnbdec = GST_AMRNBDEC (dec);
|
||||
|
||||
/* no fancy flushing */
|
||||
if (!buffer || !gst_buffer_get_size (buffer))
|
||||
return GST_FLOW_OK;
|
||||
|
||||
if (amrnbdec->rate == 0 || amrnbdec->channels == 0)
|
||||
goto not_negotiated;
|
||||
|
||||
data = gst_buffer_map (buffer, NULL, NULL, GST_MAP_READ);
|
||||
|
||||
/* get output */
|
||||
out = gst_buffer_new_and_alloc (160 * 2);
|
||||
GST_BUFFER_DURATION (out) = amrnbdec->duration;
|
||||
GST_BUFFER_TIMESTAMP (out) = amrnbdec->ts;
|
||||
|
||||
if (amrnbdec->ts != -1)
|
||||
amrnbdec->ts += amrnbdec->duration;
|
||||
if (amrnbdec->discont) {
|
||||
GST_BUFFER_FLAG_SET (out, GST_BUFFER_FLAG_DISCONT);
|
||||
amrnbdec->discont = FALSE;
|
||||
}
|
||||
|
||||
/* decode */
|
||||
out_data = gst_buffer_map (out, NULL, NULL, GST_MAP_WRITE);
|
||||
Decoder_Interface_Decode (amrnbdec->handle, data, out_data, 0);
|
||||
gst_buffer_unmap (out, out_data, -1);
|
||||
g_free (data);
|
||||
gst_buffer_unmap (out, out_data, 160 * 2);
|
||||
|
||||
/* send out */
|
||||
ret = gst_pad_push (amrnbdec->srcpad, out);
|
||||
}
|
||||
gst_buffer_unmap (buffer, data, -1);
|
||||
|
||||
gst_object_unref (amrnbdec);
|
||||
|
||||
return ret;
|
||||
return gst_audio_decoder_finish_frame (dec, out, 1);
|
||||
|
||||
/* ERRORS */
|
||||
not_negotiated:
|
||||
|
@ -402,57 +317,4 @@ not_negotiated:
|
|||
gst_object_unref (amrnbdec);
|
||||
return GST_FLOW_NOT_NEGOTIATED;
|
||||
}
|
||||
invalid_variant:
|
||||
{
|
||||
GST_ELEMENT_ERROR (amrnbdec, STREAM, TYPE_NOT_FOUND, (NULL),
|
||||
("Invalid variant"));
|
||||
gst_object_unref (amrnbdec);
|
||||
return GST_FLOW_ERROR;
|
||||
}
|
||||
}
|
||||
|
||||
static GstStateChangeReturn
|
||||
gst_amrnbdec_state_change (GstElement * element, GstStateChange transition)
|
||||
{
|
||||
GstAmrnbDec *amrnbdec;
|
||||
GstStateChangeReturn ret;
|
||||
|
||||
amrnbdec = GST_AMRNBDEC (element);
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_NULL_TO_READY:
|
||||
if (!(amrnbdec->handle = Decoder_Interface_init ()))
|
||||
goto init_failed;
|
||||
break;
|
||||
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
||||
gst_adapter_clear (amrnbdec->adapter);
|
||||
amrnbdec->rate = 0;
|
||||
amrnbdec->channels = 0;
|
||||
amrnbdec->ts = -1;
|
||||
amrnbdec->discont = TRUE;
|
||||
gst_segment_init (&amrnbdec->segment, GST_FORMAT_TIME);
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_READY_TO_NULL:
|
||||
Decoder_Interface_exit (amrnbdec->handle);
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
return ret;
|
||||
|
||||
/* ERRORS */
|
||||
init_failed:
|
||||
{
|
||||
GST_ELEMENT_ERROR (amrnbdec, LIBRARY, INIT, (NULL),
|
||||
("Failed to open AMR Decoder"));
|
||||
return GST_STATE_CHANGE_FAILURE;
|
||||
}
|
||||
}
|
||||
|
|
|
@ -21,7 +21,7 @@
|
|||
#define __GST_AMRNBDEC_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/base/gstadapter.h>
|
||||
#include <gst/audio/gstaudiodecoder.h>
|
||||
#include <interf_dec.h>
|
||||
|
||||
G_BEGIN_DECLS
|
||||
|
@ -47,29 +47,19 @@ typedef enum
|
|||
} GstAmrnbVariant;
|
||||
|
||||
struct _GstAmrnbDec {
|
||||
GstElement element;
|
||||
|
||||
/* pads */
|
||||
GstPad *sinkpad, *srcpad;
|
||||
guint64 ts;
|
||||
GstAudioDecoder element;
|
||||
|
||||
GstAmrnbVariant variant;
|
||||
|
||||
GstAdapter *adapter;
|
||||
|
||||
/* library handle */
|
||||
void *handle;
|
||||
|
||||
/* output settings */
|
||||
gint channels, rate;
|
||||
gint duration;
|
||||
|
||||
GstSegment segment;
|
||||
gboolean discont;
|
||||
};
|
||||
|
||||
struct _GstAmrnbDecClass {
|
||||
GstElementClass parent_class;
|
||||
GstAudioDecoderClass parent_class;
|
||||
};
|
||||
|
||||
GType gst_amrnbdec_get_type (void);
|
||||
|
|
Loading…
Reference in a new issue