*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
/**
* <refsect2>
* <title>Example launch line</title>
* |[
- * gst-launch filesrc location=abc.dts ! dcaparse ! dtsdec ! audioresample ! audioconvert ! autoaudiosink
+ * gst-launch-1.0 filesrc location=abc.dts ! dcaparse ! dtsdec ! audioresample ! audioconvert ! autoaudiosink
* ]|
* </refsect2>
*/
#include <string.h>
#include "gstdcaparse.h"
-#include <gst/base/gstbytereader.h>
-#include <gst/base/gstbitreader.h>
+#include <gst/base/base.h>
+#include <gst/pbutils/pbutils.h>
GST_DEBUG_CATEGORY_STATIC (dca_parse_debug);
#define GST_CAT_DEFAULT dca_parse_debug
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-dts"));
+ GST_STATIC_CAPS ("audio/x-dts; " "audio/x-private1-dts"));
static void gst_dca_parse_finalize (GObject * object);
static gboolean gst_dca_parse_stop (GstBaseParse * parse);
static GstFlowReturn gst_dca_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize);
+static GstFlowReturn gst_dca_parse_pre_push_frame (GstBaseParse * parse,
+ GstBaseParseFrame * frame);
static GstCaps *gst_dca_parse_get_sink_caps (GstBaseParse * parse,
GstCaps * filter);
+static gboolean gst_dca_parse_set_sink_caps (GstBaseParse * parse,
+ GstCaps * caps);
#define gst_dca_parse_parent_class parent_class
G_DEFINE_TYPE (GstDcaParse, gst_dca_parse, GST_TYPE_BASE_PARSE);
parse_class->start = GST_DEBUG_FUNCPTR (gst_dca_parse_start);
parse_class->stop = GST_DEBUG_FUNCPTR (gst_dca_parse_stop);
parse_class->handle_frame = GST_DEBUG_FUNCPTR (gst_dca_parse_handle_frame);
+ parse_class->pre_push_frame =
+ GST_DEBUG_FUNCPTR (gst_dca_parse_pre_push_frame);
parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_dca_parse_get_sink_caps);
+ parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_dca_parse_set_sink_caps);
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,
+ gst_element_class_set_static_metadata (element_class,
"DTS Coherent Acoustics audio stream parser", "Codec/Parser/Audio",
"DCA parser", "Tim-Philipp Müller <tim centricular net>");
}
dcaparse->block_size = -1;
dcaparse->frame_size = -1;
dcaparse->last_sync = 0;
+ dcaparse->sent_codec_tag = FALSE;
}
static void
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (dcaparse),
DCA_MIN_FRAMESIZE);
gst_dca_parse_reset (dcaparse);
+ dcaparse->baseparse_chainfunc =
+ GST_BASE_PARSE_SINK_PAD (GST_BASE_PARSE (dcaparse))->chainfunc;
+
+ GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (dcaparse));
+ GST_PAD_SET_ACCEPT_TEMPLATE (GST_BASE_PARSE_SINK_PAD (dcaparse));
}
static void
GstDcaParse *dcaparse = GST_DCA_PARSE (parse);
GstBuffer *buf = frame->buffer;
GstByteReader r;
- gboolean parser_draining;
gboolean parser_in_sync;
gboolean terminator;
guint32 sync = 0;
- guint size, rate, chans, num_blocks, samples_per_block, depth;
+ guint size = 0, rate, chans, num_blocks, samples_per_block, depth;
gint block_size;
gint endianness;
gint off = -1;
dcaparse->last_sync = sync;
+ /* FIXME: Don't look for a second syncword, there are streams out there
+ * that consistently contain garbage between every frame so we never ever
+ * find a second consecutive syncword.
+ * See https://bugzilla.gnome.org/show_bug.cgi?id=738237
+ */
+#if 0
parser_draining = GST_BASE_PARSE_DRAINING (parse);
if (!parser_in_sync && !parser_draining) {
goto cleanup;
}
}
+#endif
/* found frame */
ret = GST_FLOW_OK;
return ret;
}
+/*
+ * MPEG-PS private1 streams add a 2 bytes "Audio Substream Headers" for each
+ * buffer (not each frame) with the offset of the next frame's start.
+ * These 2 bytes can be dropped safely as they do not include any timing
+ * information, only the offset to the start of the next frame.
+ * See gstac3parse.c for a more detailed description.
+ * */
+
+static GstFlowReturn
+gst_dca_parse_chain_priv (GstPad * pad, GstObject * parent, GstBuffer * buffer)
+{
+ GstDcaParse *dcaparse = GST_DCA_PARSE (parent);
+ GstFlowReturn ret;
+ GstBuffer *newbuf;
+ gsize size;
+
+ size = gst_buffer_get_size (buffer);
+ if (size >= 2) {
+ newbuf = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, 2, size - 2);
+ gst_buffer_unref (buffer);
+ ret = dcaparse->baseparse_chainfunc (pad, parent, newbuf);
+ } else {
+ gst_buffer_unref (buffer);
+ ret = GST_FLOW_OK;
+ }
+
+ return ret;
+}
+
+static void
+remove_fields (GstCaps * caps)
+{
+ guint i, n;
+
+ n = gst_caps_get_size (caps);
+ for (i = 0; i < n; i++) {
+ GstStructure *s = gst_caps_get_structure (caps, i);
+
+ gst_structure_remove_field (s, "framed");
+ }
+}
+
static GstCaps *
gst_dca_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter)
{
GstCaps *peercaps, *templ;
GstCaps *res;
- /* FIXME: handle filter caps */
- templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse));
+ templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));
+ if (filter) {
+ GstCaps *fcopy = gst_caps_copy (filter);
+ /* Remove the fields we convert */
+ remove_fields (fcopy);
+ peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), fcopy);
+ gst_caps_unref (fcopy);
+ } else
+ peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), NULL);
- peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
if (peercaps) {
- guint i, n;
-
/* Remove the framed field */
peercaps = gst_caps_make_writable (peercaps);
- n = gst_caps_get_size (peercaps);
- for (i = 0; i < n; i++) {
- GstStructure *s = gst_caps_get_structure (peercaps, i);
-
- gst_structure_remove_field (s, "framed");
- }
+ remove_fields (peercaps);
res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
res = templ;
}
+ if (filter) {
+ GstCaps *intersection;
+
+ intersection =
+ gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
+ gst_caps_unref (res);
+ res = intersection;
+ }
+
return res;
}
+
+static gboolean
+gst_dca_parse_set_sink_caps (GstBaseParse * parse, GstCaps * caps)
+{
+ GstStructure *s;
+ GstDcaParse *dcaparse = GST_DCA_PARSE (parse);
+
+ s = gst_caps_get_structure (caps, 0);
+ if (gst_structure_has_name (s, "audio/x-private1-dts")) {
+ gst_pad_set_chain_function (parse->sinkpad, gst_dca_parse_chain_priv);
+ } else {
+ gst_pad_set_chain_function (parse->sinkpad, dcaparse->baseparse_chainfunc);
+ }
+ return TRUE;
+}
+
+static GstFlowReturn
+gst_dca_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
+{
+ GstDcaParse *dcaparse = GST_DCA_PARSE (parse);
+
+ if (!dcaparse->sent_codec_tag) {
+ GstTagList *taglist;
+ GstCaps *caps;
+
+ taglist = gst_tag_list_new_empty ();
+
+ /* codec tag */
+ caps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (parse));
+ gst_pb_utils_add_codec_description_to_tag_list (taglist,
+ GST_TAG_AUDIO_CODEC, caps);
+ gst_caps_unref (caps);
+
+ gst_base_parse_merge_tags (parse, taglist, GST_TAG_MERGE_REPLACE);
+ gst_tag_list_unref (taglist);
+
+ /* also signals the end of first-frame processing */
+ dcaparse->sent_codec_tag = TRUE;
+ }
+
+ return GST_FLOW_OK;
+}