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amrnb: port to 0.11

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This commit is contained in:
Wim Taymans 2011-10-04 17:35:40 +02:00
parent fa2da0c2cb
commit f9863d1274
3 changed files with 70 additions and 78 deletions
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3
configure.ac
View file

@ -211,8 +211,7 @@ dnl *** plug-ins to include ***
dnl Non ported plugins (non-dependant, then dependant)
dnl Make sure you have a space before and after all plugins
GST_PLUGINS_NONPORTED=" dvdsub iec958 synaesthesia xingmux \
mpegstream realmedia \
amrnb cdio dvdread twolame "
mpegstream realmedia cdio dvdread twolame "
AC_SUBST(GST_PLUGINS_NONPORTED)
dnl these are all the gst plug-ins, compilable without additional libs

87
ext/amrnb/amrnbdec.c
View file

@ -104,27 +104,8 @@ static GstStateChangeReturn gst_amrnbdec_state_change (GstElement * element,
static void gst_amrnbdec_finalize (GObject * object);
#define _do_init(bla) \
GST_DEBUG_CATEGORY_INIT (gst_amrnbdec_debug, "amrnbdec", 0, "AMR-NB audio decoder");
GST_BOILERPLATE_FULL (GstAmrnbDec, gst_amrnbdec, GstElement, GST_TYPE_ELEMENT,
_do_init);
static void
gst_amrnbdec_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template));
gst_element_class_set_details_simple (element_class, "AMR-NB audio decoder",
"Codec/Decoder/Audio",
"Adaptive Multi-Rate Narrow-Band audio decoder",
"GStreamer maintainers <gstreamer-devel@lists.sourceforge.net>");
}
#define gst_amrnbdec_parent_class parent_class
G_DEFINE_TYPE (GstAmrnbDec, gst_amrnbdec, GST_TYPE_ELEMENT);
static void
gst_amrnbdec_class_init (GstAmrnbDecClass * klass)
@ -143,14 +124,26 @@ gst_amrnbdec_class_init (GstAmrnbDecClass * klass)
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
element_class->change_state = GST_DEBUG_FUNCPTR (gst_amrnbdec_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,
gst_static_pad_template_get (&src_template));
gst_element_class_set_details_simple (element_class, "AMR-NB audio decoder",
"Codec/Decoder/Audio",
"Adaptive Multi-Rate Narrow-Band audio decoder",
"GStreamer maintainers <gstreamer-devel@lists.sourceforge.net>");
GST_DEBUG_CATEGORY_INIT (gst_amrnbdec_debug, "amrnbdec", 0,
"AMR-NB audio decoder");
}
static void
gst_amrnbdec_init (GstAmrnbDec * amrnbdec, GstAmrnbDecClass * klass)
gst_amrnbdec_init (GstAmrnbDec * amrnbdec)
{
/* create the sink pad */
amrnbdec->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
gst_pad_set_setcaps_function (amrnbdec->sinkpad, gst_amrnbdec_setcaps);
gst_pad_set_event_function (amrnbdec->sinkpad, gst_amrnbdec_event);
gst_pad_set_chain_function (amrnbdec->sinkpad, gst_amrnbdec_chain);
gst_element_add_pad (GST_ELEMENT (amrnbdec), amrnbdec->sinkpad);
@ -256,6 +249,15 @@ gst_amrnbdec_event (GstPad * pad, GstEvent * event)
amrnbdec = GST_AMRNBDEC (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
ret = gst_amrnbdec_setcaps (pad, caps);
gst_event_unref (event);
break;
}
case GST_EVENT_FLUSH_START:
ret = gst_pad_push_event (amrnbdec->srcpad, event);
break;
@ -268,32 +270,24 @@ gst_amrnbdec_event (GstPad * pad, GstEvent * event)
gst_adapter_clear (amrnbdec->adapter);
ret = gst_pad_push_event (amrnbdec->srcpad, event);
break;
case GST_EVENT_NEWSEGMENT:
case GST_EVENT_SEGMENT:
{
GstFormat format;
gdouble rate, arate;
gint64 start, stop, time;
gboolean update;
GstSegment seg;
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
gst_event_copy_segment (event, &seg);
/* we need time for now */
if (format != GST_FORMAT_TIME)
if (seg.format != GST_FORMAT_TIME)
goto newseg_wrong_format;
GST_DEBUG_OBJECT (amrnbdec,
"newsegment: update %d, rate %g, arate %g, start %" GST_TIME_FORMAT
", stop %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT,
update, rate, arate, GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
GST_TIME_ARGS (time));
GST_DEBUG_OBJECT (amrnbdec, "segment: %" GST_SEGMENT_FORMAT, &seg);
/* now configure the values */
gst_segment_set_newsegment_full (&amrnbdec->segment, update,
rate, arate, format, start, stop, time);
amrnbdec->segment = seg;
ret = gst_pad_push_event (amrnbdec->srcpad, event);
}
break;
}
default:
ret = gst_pad_push_event (amrnbdec->srcpad, event);
break;
@ -341,22 +335,24 @@ gst_amrnbdec_chain (GstPad * pad, GstBuffer * buffer)
while (TRUE) {
GstBuffer *out;
guint8 head[1];
guint8 *data;
short *out_data;
gint block, mode;
/* need to peek data to get the size */
if (gst_adapter_available (amrnbdec->adapter) < 1)
break;
data = (guint8 *) gst_adapter_peek (amrnbdec->adapter, 1);
gst_adapter_copy (amrnbdec->adapter, head, 0, 1);
/* get size */
switch (amrnbdec->variant) {
case GST_AMRNB_VARIANT_IF1:
mode = (data[0] >> 3) & 0x0F;
mode = (head[0] >> 3) & 0x0F;
block = block_size_if1[mode] + 1;
break;
case GST_AMRNB_VARIANT_IF2:
mode = data[0] & 0x0F;
mode = head[0] & 0x0F;
block = block_size_if2[mode] + 1;
break;
default:
@ -384,11 +380,10 @@ gst_amrnbdec_chain (GstPad * pad, GstBuffer * buffer)
amrnbdec->discont = FALSE;
}
gst_buffer_set_caps (out, GST_PAD_CAPS (amrnbdec->srcpad));
/* decode */
Decoder_Interface_Decode (amrnbdec->handle, data,
(short *) GST_BUFFER_DATA (out), 0);
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);
/* send out */

58
ext/amrnb/amrnbenc.c
View file

@ -99,8 +99,8 @@ static gboolean gst_amrnbenc_set_format (GstAudioEncoder * enc,
static GstFlowReturn gst_amrnbenc_handle_frame (GstAudioEncoder * enc,
GstBuffer * in_buf);
GST_BOILERPLATE (GstAmrnbEnc, gst_amrnbenc, GstAudioEncoder,
GST_TYPE_AUDIO_ENCODER);
#define gst_amrnbenc_parent_class parent_class
G_DEFINE_TYPE (GstAmrnbEnc, gst_amrnbenc, GST_TYPE_AUDIO_ENCODER);
static void
gst_amrnbenc_set_property (GObject * object, guint prop_id,
@ -136,26 +136,11 @@ gst_amrnbenc_get_property (GObject * object, guint prop_id,
return;
}
static void
gst_amrnbenc_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template));
gst_element_class_set_details_simple (element_class, "AMR-NB audio encoder",
"Codec/Encoder/Audio",
"Adaptive Multi-Rate Narrow-Band audio encoder",
"Wim Taymans <wim.taymans@gmail.com>");
}
static void
gst_amrnbenc_class_init (GstAmrnbEncClass * klass)
{
GObjectClass *object_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
object_class->set_property = gst_amrnbenc_set_property;
@ -172,12 +157,22 @@ gst_amrnbenc_class_init (GstAmrnbEncClass * klass)
BANDMODE_DEFAULT,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template));
gst_element_class_set_details_simple (element_class, "AMR-NB audio encoder",
"Codec/Encoder/Audio",
"Adaptive Multi-Rate Narrow-Band audio encoder",
"Wim Taymans <wim.taymans@gmail.com>");
GST_DEBUG_CATEGORY_INIT (gst_amrnbenc_debug, "amrnbenc", 0,
"AMR-NB audio encoder");
}
static void
gst_amrnbenc_init (GstAmrnbEnc * amrnbenc, GstAmrnbEncClass * klass)
gst_amrnbenc_init (GstAmrnbEnc * amrnbenc)
{
}
@ -249,8 +244,9 @@ gst_amrnbenc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
GstAmrnbEnc *amrnbenc;
GstFlowReturn ret;
GstBuffer *out;
guint8 *data;
gint outsize;
short *in_data;
guint8 *out_data;
gsize in_size, out_size;
amrnbenc = GST_AMRNBENC (enc);
@ -262,9 +258,11 @@ gst_amrnbenc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
return GST_FLOW_OK;
}
if (G_UNLIKELY (GST_BUFFER_SIZE (buffer) < 320)) {
GST_DEBUG_OBJECT (amrnbenc, "discarding trailing data %d",
buffer ? GST_BUFFER_SIZE (buffer) : 0);
in_data = gst_buffer_map (buffer, &in_size, NULL, GST_MAP_READ);
if (G_UNLIKELY (in_size < 320)) {
gst_buffer_unmap (buffer, in_data, in_size);
GST_DEBUG_OBJECT (amrnbenc, "discarding trailing data %d", in_size);
return gst_audio_encoder_finish_frame (enc, NULL, -1);
}
@ -272,17 +270,17 @@ gst_amrnbenc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
out = gst_buffer_new_and_alloc (32);
/* AMR encoder actually writes into the source data buffers it gets */
/* should be able to handle that with what we are given */
data = GST_BUFFER_DATA (buffer);
out_data = gst_buffer_map (buffer, NULL, NULL, GST_MAP_WRITE);
/* encode */
outsize =
out_size =
Encoder_Interface_Encode (amrnbenc->handle, amrnbenc->bandmode,
(short *) data, (guint8 *) GST_BUFFER_DATA (out), 0);
in_data, out_data, 0);
gst_buffer_unmap (out, out_data, out_size);
GST_LOG_OBJECT (amrnbenc, "output data size %d", outsize);
GST_LOG_OBJECT (amrnbenc, "output data size %d", out_size);
if (outsize) {
GST_BUFFER_SIZE (out) = outsize;
if (out_size) {
ret = gst_audio_encoder_finish_frame (enc, out, 160);
} else {
/* should not happen (without dtx or so at least) */