mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-24 01:00:37 +00:00
Merge branch 'master' into 0.11
Conflicts: ext/amrwbdec/Makefile.am ext/amrwbdec/amrwbdec.c
This commit is contained in:
Wim Taymans
commit
f99001a05d
5 changed files with 106 additions and 247 deletions
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@ -7,8 +7,9 @@ libgstamrnb_la_SOURCES = \
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libgstamrnb_la_CFLAGS = -DGST_USE_UNSTABLE_API $(GST_PLUGINS_BASE_CFLAGS) \
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$(GST_BASE_CFLAGS) $(GST_CFLAGS) $(AMRNB_CFLAGS)
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libgstamrnb_la_LIBADD = $(GST_BASE_LIBS) -lgstaudio-@GST_MAJORMINOR@ \
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$(GST_LIBS) $(AMRNB_LIBS)
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libgstamrnb_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) \
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-lgstaudio-@GST_MAJORMINOR@ \
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$(GST_BASE_LIBS) $(GST_LIBS) $(AMRNB_LIBS)
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libgstamrnb_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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libgstamrnb_la_LIBTOOLFLAGS = --tag=disable-static
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@ -314,7 +314,6 @@ not_negotiated:
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{
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GST_ELEMENT_ERROR (amrnbdec, STREAM, TYPE_NOT_FOUND, (NULL),
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("Decoder is not initialized"));
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gst_object_unref (amrnbdec);
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return GST_FLOW_NOT_NEGOTIATED;
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}
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}
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@ -4,8 +4,11 @@ libgstamrwbdec_la_SOURCES = \
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amrwb.c \
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amrwbdec.c
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libgstamrwbdec_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(AMRWB_CFLAGS)
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libgstamrwbdec_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(GST_LIBS) $(AMRWB_LIBS)
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libgstamrwbdec_la_CFLAGS = -DGST_USE_UNSTABLE_API $(GST_PLUGINS_BASE_CFLAGS) \
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$(GST_CFLAGS) $(AMRWB_CFLAGS)
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libgstamrwbdec_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) \
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-lgstaudio-@GST_MAJORMINOR@ \
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$(GST_LIBS) $(AMRWB_LIBS)
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libgstamrwbdec_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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libgstamrwbdec_la_LIBTOOLFLAGS = --tag=disable-static
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@ -65,27 +65,23 @@ static const unsigned char block_size[16] =
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6, 0, 0, 0, 0, 1, 1
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};
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static gboolean gst_amrwbdec_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_amrwbdec_chain (GstPad * pad, GstBuffer * buffer);
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static gboolean gst_amrwbdec_setcaps (GstPad * pad, GstCaps * caps);
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static GstStateChangeReturn gst_amrwbdec_state_change (GstElement * element,
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GstStateChange transition);
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static void gst_amrwbdec_finalize (GObject * object);
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static gboolean gst_amrwbdec_start (GstAudioDecoder * dec);
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static gboolean gst_amrwbdec_stop (GstAudioDecoder * dec);
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static gboolean gst_amrwbdec_set_format (GstAudioDecoder * dec, GstCaps * caps);
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static gboolean gst_amrwbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
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gint * offset, gint * length);
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static GstFlowReturn gst_amrwbdec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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#define gst_amrwbdec_parent_class parent_class
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G_DEFINE_TYPE (GstAmrwbDec, gst_amrwbdec, GST_TYPE_ELEMENT);
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G_DEFINE_TYPE (GstAmrwbDec, gst_amrwbdec, GST_TYPE_AUDIO_DECODER);
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static void
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gst_amrwbdec_class_init (GstAmrwbDecClass * klass)
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{
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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object_class->finalize = gst_amrwbdec_finalize;
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element_class->change_state = GST_DEBUG_FUNCPTR (gst_amrwbdec_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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gst_element_class_add_pad_template (element_class,
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@ -96,6 +92,12 @@ gst_amrwbdec_class_init (GstAmrwbDecClass * klass)
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"Adaptive Multi-Rate Wideband audio decoder",
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"Renato Araujo <renato.filho@indt.org.br>");
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base_class->start = GST_DEBUG_FUNCPTR (gst_amrwbdec_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_amrwbdec_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrwbdec_set_format);
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base_class->parse = GST_DEBUG_FUNCPTR (gst_amrwbdec_parse);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrwbdec_handle_frame);
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GST_DEBUG_CATEGORY_INIT (gst_amrwbdec_debug, "amrwbdec", 0,
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"AMR-WB audio decoder");
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}
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@ -103,48 +105,42 @@ gst_amrwbdec_class_init (GstAmrwbDecClass * klass)
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static void
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gst_amrwbdec_init (GstAmrwbDec * amrwbdec)
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{
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/* create the sink pad */
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amrwbdec->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
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gst_pad_set_event_function (amrwbdec->sinkpad, gst_amrwbdec_event);
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gst_pad_set_chain_function (amrwbdec->sinkpad, gst_amrwbdec_chain);
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gst_element_add_pad (GST_ELEMENT (amrwbdec), amrwbdec->sinkpad);
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/* create the src pad */
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amrwbdec->srcpad = gst_pad_new_from_static_template (&src_template, "src");
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gst_pad_use_fixed_caps (amrwbdec->srcpad);
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gst_element_add_pad (GST_ELEMENT (amrwbdec), amrwbdec->srcpad);
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amrwbdec->adapter = gst_adapter_new ();
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/* init rest */
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amrwbdec->handle = NULL;
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amrwbdec->channels = 0;
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amrwbdec->rate = 0;
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amrwbdec->duration = 0;
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amrwbdec->ts = -1;
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}
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static void
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gst_amrwbdec_finalize (GObject * object)
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{
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GstAmrwbDec *amrwbdec;
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amrwbdec = GST_AMRWBDEC (object);
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gst_adapter_clear (amrwbdec->adapter);
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g_object_unref (amrwbdec->adapter);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_amrwbdec_setcaps (GstPad * pad, GstCaps * caps)
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gst_amrwbdec_start (GstAudioDecoder * dec)
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{
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GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
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GST_DEBUG_OBJECT (dec, "start");
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if (!(amrwbdec->handle = D_IF_init ()))
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return FALSE;
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amrwbdec->rate = 0;
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amrwbdec->channels = 0;
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return TRUE;
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}
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static gboolean
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gst_amrwbdec_stop (GstAudioDecoder * dec)
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{
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GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
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GST_DEBUG_OBJECT (dec, "stop");
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D_IF_exit (amrwbdec->handle);
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return TRUE;
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}
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static gboolean
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gst_amrwbdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
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{
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GstStructure *structure;
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GstAmrwbDec *amrwbdec;
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GstCaps *copy;
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amrwbdec = GST_AMRWBDEC (gst_pad_get_parent (pad));
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amrwbdec = GST_AMRWBDEC (dec);
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structure = gst_caps_get_structure (caps, 0);
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@ -153,224 +149,94 @@ gst_amrwbdec_setcaps (GstPad * pad, GstCaps * caps)
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gst_structure_get_int (structure, "rate", &amrwbdec->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, amrwbdec->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, amrwbdec->rate, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
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"rate", G_TYPE_INT, amrwbdec->rate, NULL);
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amrwbdec->duration = gst_util_uint64_scale_int (GST_SECOND, L_FRAME16k,
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amrwbdec->rate * amrwbdec->channels);
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gst_pad_set_caps (amrwbdec->srcpad, copy);
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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 (amrwbdec);
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return TRUE;
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}
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static gboolean
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gst_amrwbdec_event (GstPad * pad, GstEvent * event)
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static GstFlowReturn
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gst_amrwbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
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gint * offset, gint * length)
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{
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GstAmrwbDec *amrwbdec;
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gboolean ret = TRUE;
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GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
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guint8 header[1];
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guint size;
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gboolean sync, eos;
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gint block, mode;
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amrwbdec = GST_AMRWBDEC (gst_pad_get_parent (pad));
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size = gst_adapter_available (adapter);
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g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
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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_audio_decoder_get_parse_state (dec, &sync, &eos);
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gst_event_parse_caps (event, &caps);
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ret = gst_amrwbdec_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 (amrwbdec->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 (amrwbdec->srcpad, event);
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gst_adapter_clear (amrwbdec->adapter);
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amrwbdec->ts = -1;
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break;
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case GST_EVENT_EOS:
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gst_adapter_clear (amrwbdec->adapter);
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ret = gst_pad_push_event (amrwbdec->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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/* need to peek data to get the size */
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if (gst_adapter_available (adapter) < 1)
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return GST_FLOW_ERROR;
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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_adapter_copy (adapter, header, 0, 1);
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mode = (header[0] >> 3) & 0x0F;
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block = block_size[mode];
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GST_DEBUG_OBJECT (amrwbdec, "segment: %" GST_SEGMENT_FORMAT, &seg);
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GST_DEBUG_OBJECT (amrwbdec, "mode %d, block %d", mode, block);
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/* now configure the values */
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amrwbdec->segment = seg;
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ret = gst_pad_push_event (amrwbdec->srcpad, event);
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}
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break;
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default:
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ret = gst_pad_push_event (amrwbdec->srcpad, event);
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break;
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if (block) {
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*offset = 0;
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*length = block;
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} else {
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/* no frame yet, skip one byte */
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GST_LOG_OBJECT (amrwbdec, "skipping byte");
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*offset = 1;
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return GST_FLOW_UNEXPECTED;
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}
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done:
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gst_object_unref (amrwbdec);
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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 (amrwbdec, "received non TIME newsegment");
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goto done;
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}
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return GST_FLOW_OK;
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}
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static GstFlowReturn
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gst_amrwbdec_chain (GstPad * pad, GstBuffer * buffer)
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gst_amrwbdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
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{
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GstAmrwbDec *amrwbdec;
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GstFlowReturn ret = GST_FLOW_OK;
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GstBuffer *out;
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guint8 *data;
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Word16 *outdata;
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amrwbdec = GST_AMRWBDEC (gst_pad_get_parent (pad));
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amrwbdec = GST_AMRWBDEC (dec);
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/* no fancy flushing */
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if (!buffer || !gst_buffer_get_size (buffer))
|
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return GST_FLOW_OK;
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|
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if (amrwbdec->rate == 0 || amrwbdec->channels == 0)
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goto not_negotiated;
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|
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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 (amrwbdec->adapter);
|
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amrwbdec->ts = -1;
|
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amrwbdec->discont = TRUE;
|
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}
|
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/* the library seems to write into the source data, hence the copy. */
|
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/* should be no problem */
|
||||
data = gst_buffer_map (buffer, NULL, NULL, GST_MAP_READ);
|
||||
|
||||
/* 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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amrwbdec->ts = GST_BUFFER_TIMESTAMP (buffer);
|
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/* get output */
|
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out = gst_buffer_new_and_alloc (sizeof (gint16) * L_FRAME16k);
|
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outdata = gst_buffer_map (out, NULL, NULL, GST_MAP_WRITE);
|
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|
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gst_adapter_push (amrwbdec->adapter, buffer);
|
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/* decode */
|
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D_IF_decode (amrwbdec->handle, (unsigned char *) data, outdata, _good_frame);
|
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|
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while (TRUE) {
|
||||
GstBuffer *out;
|
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guint8 head[1];
|
||||
const guint8 *data;
|
||||
Word16 *out_data;
|
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gint block, mode;
|
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gst_buffer_unmap (out, outdata, sizeof (gint16) * L_FRAME16k);
|
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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");
|
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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:
|
||||
{
|
||||
GST_ELEMENT_ERROR (amrwbdec, STREAM, TYPE_NOT_FOUND, (NULL),
|
||||
("Decoder is not initialized"));
|
||||
gst_object_unref (amrwbdec);
|
||||
return GST_FLOW_NOT_NEGOTIATED;
|
||||
}
|
||||
}
|
||||
|
||||
static GstStateChangeReturn
|
||||
gst_amrwbdec_state_change (GstElement * element, GstStateChange transition)
|
||||
{
|
||||
GstAmrwbDec *amrwbdec;
|
||||
GstStateChangeReturn ret;
|
||||
|
||||
amrwbdec = GST_AMRWBDEC (element);
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_NULL_TO_READY:
|
||||
if (!(amrwbdec->handle = D_IF_init ()))
|
||||
goto init_failed;
|
||||
break;
|
||||
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
||||
gst_adapter_clear (amrwbdec->adapter);
|
||||
amrwbdec->rate = 0;
|
||||
amrwbdec->channels = 0;
|
||||
amrwbdec->ts = -1;
|
||||
amrwbdec->discont = TRUE;
|
||||
gst_segment_init (&amrwbdec->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:
|
||||
D_IF_exit (amrwbdec->handle);
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
return ret;
|
||||
|
||||
/* ERRORS */
|
||||
init_failed:
|
||||
{
|
||||
GST_ELEMENT_ERROR (amrwbdec, LIBRARY, INIT, (NULL),
|
||||
("Failed to open AMR Decoder"));
|
||||
return GST_STATE_CHANGE_FAILURE;
|
||||
}
|
||||
}
|
||||
|
|
|
|||
|
|
@ -21,7 +21,7 @@
|
|||
#define __GST_AMRWBDEC_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/base/gstadapter.h>
|
||||
#include <gst/audio/gstaudiodecoder.h>
|
||||
#include <dec_if.h>
|
||||
#include <if_rom.h>
|
||||
|
||||
|
|
@ -47,27 +47,17 @@ typedef struct _GstAmrwbDecClass GstAmrwbDecClass;
|
|||
* Opaque data structure.
|
||||
*/
|
||||
struct _GstAmrwbDec {
|
||||
GstElement element;
|
||||
|
||||
/* pads */
|
||||
GstPad *sinkpad, *srcpad;
|
||||
guint64 ts;
|
||||
|
||||
GstAdapter *adapter;
|
||||
GstAudioDecoder element;
|
||||
|
||||
/* library handle */
|
||||
void *handle;
|
||||
|
||||
/* output settings */
|
||||
gint channels, rate;
|
||||
gint duration;
|
||||
|
||||
GstSegment segment;
|
||||
gboolean discont;
|
||||
};
|
||||
|
||||
struct _GstAmrwbDecClass {
|
||||
GstElementClass parent_class;
|
||||
GstAudioDecoderClass parent_class;
|
||||
};
|
||||
|
||||
GType gst_amrwbdec_get_type (void);
|
||||
|
|
|
|||
Loading…
Reference in a new issue