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amrnb: port to 0.11
This commit is contained in:
Wim Taymans
parent
fa2da0c2cb
commit
f9863d1274
3 changed files with 70 additions and 78 deletions
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@ -211,8 +211,7 @@ dnl *** plug-ins to include ***
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dnl Non ported plugins (non-dependant, then dependant)
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dnl Make sure you have a space before and after all plugins
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GST_PLUGINS_NONPORTED=" dvdsub iec958 synaesthesia xingmux \
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mpegstream realmedia \
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amrnb cdio dvdread twolame "
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mpegstream realmedia cdio dvdread twolame "
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AC_SUBST(GST_PLUGINS_NONPORTED)
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dnl these are all the gst plug-ins, compilable without additional libs
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@ -104,27 +104,8 @@ static GstStateChangeReturn gst_amrnbdec_state_change (GstElement * element,
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static void gst_amrnbdec_finalize (GObject * object);
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#define _do_init(bla) \
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GST_DEBUG_CATEGORY_INIT (gst_amrnbdec_debug, "amrnbdec", 0, "AMR-NB audio decoder");
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GST_BOILERPLATE_FULL (GstAmrnbDec, gst_amrnbdec, GstElement, GST_TYPE_ELEMENT,
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_do_init);
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static void
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gst_amrnbdec_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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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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gst_static_pad_template_get (&src_template));
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gst_element_class_set_details_simple (element_class, "AMR-NB audio decoder",
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"Codec/Decoder/Audio",
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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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}
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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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static void
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gst_amrnbdec_class_init (GstAmrnbDecClass * klass)
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@ -143,14 +124,26 @@ gst_amrnbdec_class_init (GstAmrnbDecClass * klass)
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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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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_set_details_simple (element_class, "AMR-NB audio decoder",
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"Codec/Decoder/Audio",
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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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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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static void
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gst_amrnbdec_init (GstAmrnbDec * amrnbdec, GstAmrnbDecClass * klass)
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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_setcaps_function (amrnbdec->sinkpad, gst_amrnbdec_setcaps);
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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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@ -256,6 +249,15 @@ gst_amrnbdec_event (GstPad * pad, GstEvent * event)
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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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@ -268,32 +270,24 @@ gst_amrnbdec_event (GstPad * pad, GstEvent * event)
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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_NEWSEGMENT:
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case GST_EVENT_SEGMENT:
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{
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GstFormat format;
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gdouble rate, arate;
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gint64 start, stop, time;
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gboolean update;
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GstSegment seg;
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gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
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&start, &stop, &time);
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gst_event_copy_segment (event, &seg);
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/* we need time for now */
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if (format != GST_FORMAT_TIME)
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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,
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"newsegment: update %d, rate %g, arate %g, start %" GST_TIME_FORMAT
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", stop %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT,
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update, rate, arate, GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
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GST_TIME_ARGS (time));
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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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gst_segment_set_newsegment_full (&amrnbdec->segment, update,
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rate, arate, format, start, stop, time);
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amrnbdec->segment = seg;
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ret = gst_pad_push_event (amrnbdec->srcpad, event);
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}
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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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@ -341,22 +335,24 @@ gst_amrnbdec_chain (GstPad * pad, GstBuffer * buffer)
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while (TRUE) {
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GstBuffer *out;
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guint8 head[1];
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guint8 *data;
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short *out_data;
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gint block, mode;
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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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data = (guint8 *) gst_adapter_peek (amrnbdec->adapter, 1);
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gst_adapter_copy (amrnbdec->adapter, head, 0, 1);
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/* get size */
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switch (amrnbdec->variant) {
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case GST_AMRNB_VARIANT_IF1:
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mode = (data[0] >> 3) & 0x0F;
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mode = (head[0] >> 3) & 0x0F;
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block = block_size_if1[mode] + 1;
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break;
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case GST_AMRNB_VARIANT_IF2:
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mode = data[0] & 0x0F;
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mode = head[0] & 0x0F;
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block = block_size_if2[mode] + 1;
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break;
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default:
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@ -384,11 +380,10 @@ gst_amrnbdec_chain (GstPad * pad, GstBuffer * buffer)
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amrnbdec->discont = FALSE;
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}
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gst_buffer_set_caps (out, GST_PAD_CAPS (amrnbdec->srcpad));
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/* decode */
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Decoder_Interface_Decode (amrnbdec->handle, data,
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(short *) GST_BUFFER_DATA (out), 0);
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out_data = gst_buffer_map (out, NULL, NULL, GST_MAP_WRITE);
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Decoder_Interface_Decode (amrnbdec->handle, data, out_data, 0);
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gst_buffer_unmap (out, out_data, -1);
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g_free (data);
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/* send out */
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@ -99,8 +99,8 @@ static gboolean gst_amrnbenc_set_format (GstAudioEncoder * enc,
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static GstFlowReturn gst_amrnbenc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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GST_BOILERPLATE (GstAmrnbEnc, gst_amrnbenc, GstAudioEncoder,
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GST_TYPE_AUDIO_ENCODER);
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#define gst_amrnbenc_parent_class parent_class
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G_DEFINE_TYPE (GstAmrnbEnc, gst_amrnbenc, GST_TYPE_AUDIO_ENCODER);
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static void
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gst_amrnbenc_set_property (GObject * object, guint prop_id,
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@ -136,26 +136,11 @@ gst_amrnbenc_get_property (GObject * object, guint prop_id,
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return;
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}
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static void
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gst_amrnbenc_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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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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gst_static_pad_template_get (&src_template));
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gst_element_class_set_details_simple (element_class, "AMR-NB audio encoder",
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"Codec/Encoder/Audio",
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"Adaptive Multi-Rate Narrow-Band audio encoder",
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"Wim Taymans <wim.taymans@gmail.com>");
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}
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static void
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gst_amrnbenc_class_init (GstAmrnbEncClass * 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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GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
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object_class->set_property = gst_amrnbenc_set_property;
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@ -172,12 +157,22 @@ gst_amrnbenc_class_init (GstAmrnbEncClass * klass)
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BANDMODE_DEFAULT,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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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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gst_static_pad_template_get (&src_template));
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gst_element_class_set_details_simple (element_class, "AMR-NB audio encoder",
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"Codec/Encoder/Audio",
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"Adaptive Multi-Rate Narrow-Band audio encoder",
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"Wim Taymans <wim.taymans@gmail.com>");
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GST_DEBUG_CATEGORY_INIT (gst_amrnbenc_debug, "amrnbenc", 0,
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"AMR-NB audio encoder");
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}
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static void
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gst_amrnbenc_init (GstAmrnbEnc * amrnbenc, GstAmrnbEncClass * klass)
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gst_amrnbenc_init (GstAmrnbEnc * amrnbenc)
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{
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}
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@ -249,8 +244,9 @@ gst_amrnbenc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
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GstAmrnbEnc *amrnbenc;
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GstFlowReturn ret;
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GstBuffer *out;
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guint8 *data;
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gint outsize;
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short *in_data;
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guint8 *out_data;
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gsize in_size, out_size;
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amrnbenc = GST_AMRNBENC (enc);
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@ -262,9 +258,11 @@ gst_amrnbenc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
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return GST_FLOW_OK;
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}
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if (G_UNLIKELY (GST_BUFFER_SIZE (buffer) < 320)) {
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GST_DEBUG_OBJECT (amrnbenc, "discarding trailing data %d",
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buffer ? GST_BUFFER_SIZE (buffer) : 0);
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in_data = gst_buffer_map (buffer, &in_size, NULL, GST_MAP_READ);
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if (G_UNLIKELY (in_size < 320)) {
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gst_buffer_unmap (buffer, in_data, in_size);
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GST_DEBUG_OBJECT (amrnbenc, "discarding trailing data %d", in_size);
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return gst_audio_encoder_finish_frame (enc, NULL, -1);
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}
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@ -272,17 +270,17 @@ gst_amrnbenc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
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out = gst_buffer_new_and_alloc (32);
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/* AMR encoder actually writes into the source data buffers it gets */
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/* should be able to handle that with what we are given */
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data = GST_BUFFER_DATA (buffer);
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out_data = gst_buffer_map (buffer, NULL, NULL, GST_MAP_WRITE);
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/* encode */
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outsize =
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out_size =
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Encoder_Interface_Encode (amrnbenc->handle, amrnbenc->bandmode,
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(short *) data, (guint8 *) GST_BUFFER_DATA (out), 0);
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in_data, out_data, 0);
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gst_buffer_unmap (out, out_data, out_size);
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GST_LOG_OBJECT (amrnbenc, "output data size %d", outsize);
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GST_LOG_OBJECT (amrnbenc, "output data size %d", out_size);
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if (outsize) {
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GST_BUFFER_SIZE (out) = outsize;
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if (out_size) {
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ret = gst_audio_encoder_finish_frame (enc, out, 160);
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} else {
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/* should not happen (without dtx or so at least) */
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