gboolean gst_aac_parse_event (GstBaseParse * parse, GstEvent * event);
-#define _do_init(bla) \
- GST_DEBUG_CATEGORY_INIT (aacparse_debug, "aacparse", 0, \
- "AAC audio stream parser");
-
-GST_BOILERPLATE_FULL (GstAacParse, gst_aac_parse, GstBaseParse,
- GST_TYPE_BASE_PARSE, _do_init);
+G_DEFINE_TYPE (GstAacParse, gst_aac_parse, GST_TYPE_BASE_PARSE);
static inline gint
gst_aac_parse_get_sample_rate_from_index (guint sr_idx)
}
/**
- * gst_aac_parse_base_init:
- * @klass: #GstElementClass.
+ * gst_aac_parse_class_init:
+ * @klass: #GstAacParseClass.
*
*/
static void
-gst_aac_parse_base_init (gpointer klass)
+gst_aac_parse_class_init (GstAacParseClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
+
+ GST_DEBUG_CATEGORY_INIT (aacparse_debug, "aacparse", 0,
+ "AAC audio stream parser");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_set_details_simple (element_class,
"AAC audio stream parser", "Codec/Parser/Audio",
"Advanced Audio Coding parser", "Stefan Kost <stefan.kost@nokia.com>");
-}
-
-
-/**
- * gst_aac_parse_class_init:
- * @klass: #GstAacParseClass.
- *
- */
-static void
-gst_aac_parse_class_init (GstAacParseClass * klass)
-{
- GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
parse_class->start = GST_DEBUG_FUNCPTR (gst_aac_parse_start);
parse_class->stop = GST_DEBUG_FUNCPTR (gst_aac_parse_stop);
*
*/
static void
-gst_aac_parse_init (GstAacParse * aacparse, GstAacParseClass * klass)
+gst_aac_parse_init (GstAacParse * aacparse)
{
GST_DEBUG ("initialized");
}
GstBuffer *buf = gst_value_get_buffer (value);
if (buf) {
- const guint8 *buffer = GST_BUFFER_DATA (buf);
+ guint8 *data;
+ gsize size;
guint sr_idx;
- sr_idx = ((buffer[0] & 0x07) << 1) | ((buffer[1] & 0x80) >> 7);
- aacparse->object_type = (buffer[0] & 0xf8) >> 3;
+ data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
+
+ sr_idx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7);
+ aacparse->object_type = (data[0] & 0xf8) >> 3;
aacparse->sample_rate = gst_aac_parse_get_sample_rate_from_index (sr_idx);
- aacparse->channels = (buffer[1] & 0x78) >> 3;
+ aacparse->channels = (data[1] & 0x78) >> 3;
aacparse->header_type = DSPAAC_HEADER_NONE;
aacparse->mpegversion = 4;
+ gst_buffer_unmap (buf, data, size);
GST_DEBUG ("codec_data: object_type=%d, sample_rate=%d, channels=%d",
aacparse->object_type, aacparse->sample_rate, aacparse->channels);
gst_aac_parse_check_valid_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
{
- const guint8 *data;
+ guint8 *data;
+ gsize size;
GstAacParse *aacparse;
gboolean ret = FALSE;
gboolean lost_sync;
aacparse = GST_AAC_PARSE (parse);
buffer = frame->buffer;
- data = GST_BUFFER_DATA (buffer);
+
+ data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
if (aacparse->header_type == DSPAAC_HEADER_ADIF ||
aacparse->header_type == DSPAAC_HEADER_NONE) {
/* There is nothing to parse */
- *framesize = GST_BUFFER_SIZE (buffer);
+ *framesize = size;
ret = TRUE;
} else if (aacparse->header_type == DSPAAC_HEADER_NOT_PARSED || lost_sync) {
- ret = gst_aac_parse_detect_stream (aacparse, data, GST_BUFFER_SIZE (buffer),
+ ret = gst_aac_parse_detect_stream (aacparse, data, size,
GST_BASE_PARSE_DRAINING (parse), framesize, skipsize);
} else if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
guint needed_data = 1024;
- ret = gst_aac_parse_check_adts_frame (aacparse, data,
- GST_BUFFER_SIZE (buffer), GST_BASE_PARSE_DRAINING (parse),
- framesize, &needed_data);
+ ret = gst_aac_parse_check_adts_frame (aacparse, data, size,
+ GST_BASE_PARSE_DRAINING (parse), framesize, &needed_data);
if (!ret) {
GST_DEBUG ("buffer didn't contain valid frame");
GST_DEBUG ("buffer didn't contain valid frame");
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 1024);
}
+ gst_buffer_unmap (buffer, data, size);
return ret;
}
GstBuffer *buffer;
GstFlowReturn ret = GST_FLOW_OK;
gint rate, channels;
+ guint8 *data;
+ gsize size;
aacparse = GST_AAC_PARSE (parse);
buffer = frame->buffer;
/* see above */
frame->overhead = 7;
- gst_aac_parse_parse_adts_header (aacparse, GST_BUFFER_DATA (buffer),
+ data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
+ gst_aac_parse_parse_adts_header (aacparse, data,
&rate, &channels, NULL, NULL);
+ gst_buffer_unmap (buffer, data, size);
+
GST_LOG_OBJECT (aacparse, "rate: %d, chans: %d", rate, channels);
if (G_UNLIKELY (rate != aacparse->sample_rate
static GstFlowReturn gst_ac3_parse_parse_frame (GstBaseParse * parse,
GstBaseParseFrame * frame);
-GST_BOILERPLATE (GstAc3Parse, gst_ac3_parse, GstBaseParse, GST_TYPE_BASE_PARSE);
+#define gst_ac3_parse_parent_class parent_class
+G_DEFINE_TYPE (GstAc3Parse, gst_ac3_parse, GST_TYPE_BASE_PARSE);
static void
-gst_ac3_parse_base_init (gpointer klass)
+gst_ac3_parse_class_init (GstAc3ParseClass * klass)
{
+ GObjectClass *object_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
+
+ GST_DEBUG_CATEGORY_INIT (ac3_parse_debug, "ac3parse", 0,
+ "AC3 audio stream parser");
+
+ object_class->finalize = gst_ac3_parse_finalize;
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_set_details_simple (element_class,
"AC3 audio stream parser", "Codec/Parser/Audio",
"AC3 parser", "Tim-Philipp Müller <tim centricular net>");
-}
-
-static void
-gst_ac3_parse_class_init (GstAc3ParseClass * klass)
-{
- GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
- GObjectClass *object_class = G_OBJECT_CLASS (klass);
-
- GST_DEBUG_CATEGORY_INIT (ac3_parse_debug, "ac3parse", 0,
- "AC3 audio stream parser");
-
- object_class->finalize = gst_ac3_parse_finalize;
parse_class->start = GST_DEBUG_FUNCPTR (gst_ac3_parse_start);
parse_class->stop = GST_DEBUG_FUNCPTR (gst_ac3_parse_stop);
}
static void
-gst_ac3_parse_init (GstAc3Parse * ac3parse, GstAc3ParseClass * klass)
+gst_ac3_parse_init (GstAc3Parse * ac3parse)
{
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (ac3parse), 64 * 2);
gst_ac3_parse_reset (ac3parse);
GstFlowReturn gst_amr_parse_parse_frame (GstBaseParse * parse,
GstBaseParseFrame * frame);
-#define _do_init(bla) \
- GST_DEBUG_CATEGORY_INIT (amrparse_debug, "amrparse", 0, \
- "AMR-NB audio stream parser");
-
-GST_BOILERPLATE_FULL (GstAmrParse, gst_amr_parse, GstBaseParse,
- GST_TYPE_BASE_PARSE, _do_init);
-
+G_DEFINE_TYPE (GstAmrParse, gst_amr_parse, GST_TYPE_BASE_PARSE);
/**
- * gst_amr_parse_base_init:
- * @klass: #GstElementClass.
+ * gst_amr_parse_class_init:
+ * @klass: GstAmrParseClass.
*
*/
static void
-gst_amr_parse_base_init (gpointer klass)
+gst_amr_parse_class_init (GstAmrParseClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
+
+ GST_DEBUG_CATEGORY_INIT (amrparse_debug, "amrparse", 0,
+ "AMR-NB audio stream parser");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
"AMR audio stream parser", "Codec/Parser/Audio",
"Adaptive Multi-Rate audio parser",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
-}
-
-
-/**
- * gst_amr_parse_class_init:
- * @klass: GstAmrParseClass.
- *
- */
-static void
-gst_amr_parse_class_init (GstAmrParseClass * klass)
-{
- GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
parse_class->start = GST_DEBUG_FUNCPTR (gst_amr_parse_start);
parse_class->stop = GST_DEBUG_FUNCPTR (gst_amr_parse_stop);
*
*/
static void
-gst_amr_parse_init (GstAmrParse * amrparse, GstAmrParseClass * klass)
+gst_amr_parse_init (GstAmrParse * amrparse)
{
/* init rest */
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (amrparse), 62);
GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
{
GstBuffer *buffer;
- const guint8 *data;
+ guint8 *data;
+ gsize size;
gint fsize, mode, dsize;
GstAmrParse *amrparse;
+ gboolean ret = FALSE;
amrparse = GST_AMR_PARSE (parse);
buffer = frame->buffer;
- data = GST_BUFFER_DATA (buffer);
- dsize = GST_BUFFER_SIZE (buffer);
+
+ data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
+ dsize = size;
GST_LOG ("buffer: %d bytes", dsize);
}
/* We return FALSE, so this frame won't get pushed forward. Instead,
the "skip" value is set, so next time we will receive a valid frame. */
- return FALSE;
+ goto done;
}
/* Does this look like a possible frame header candidate? */
(!GST_BASE_PARSE_LOST_SYNC (parse) || GST_BASE_PARSE_DRAINING (parse)
|| (dsize > fsize && (data[fsize] & 0x83) == 0))) {
*framesize = fsize;
- return TRUE;
+ ret = TRUE;
+ goto done;
}
}
-
GST_LOG ("sync lost");
- return FALSE;
+
+done:
+ gst_buffer_unmap (buffer, data, size);
+
+ return ret;
}
projects / platform / upstream / gst-plugins-good.git / commitdiff