X-Git-Url: http://review.tizen.org/git/?a=blobdiff_plain;f=gst%2Frtp%2Fgstrtpac3pay.c;h=1feccedf4ae154b3dd714e1b9bcb29aaf3f364d0;hb=775ccdf9775bd7051929b257444b3be915e88ec6;hp=851b92d680f82e1a3fd489d1ada217fe2ee96483;hpb=221ea2fc444944fc1f8e0e37fdbbf6bcaf96c0bd;p=platform%2Fupstream%2Fgst-plugins-good.git diff --git a/gst/rtp/gstrtpac3pay.c b/gst/rtp/gstrtpac3pay.c index 851b92d..1fecced 100644 --- a/gst/rtp/gstrtpac3pay.c +++ b/gst/rtp/gstrtpac3pay.c @@ -13,8 +13,24 @@ * * 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. + */ + +/** + * SECTION:element-rtpac3pay + * @title: rtpac3pay + * @see_also: rtpac3depay + * + * Payload AC3 audio into RTP packets according to RFC 4184. + * For detailed information see: http://www.rfc-editor.org/rfc/rfc4184.txt + * + * ## Example pipeline + * |[ + * gst-launch-1.0 -v audiotestsrc ! avenc_ac3 ! rtpac3pay ! udpsink + * ]| This example pipeline will encode and payload AC3 stream. Refer to + * the rtpac3depay example to depayload and decode the RTP stream. + * */ #ifdef HAVE_CONFIG_H @@ -24,8 +40,10 @@ #include #include +#include #include "gstrtpac3pay.h" +#include "gstrtputils.h" GST_DEBUG_CATEGORY_STATIC (rtpac3pay_debug); #define GST_CAT_DEFAULT (rtpac3pay_debug) @@ -53,48 +71,48 @@ static void gst_rtp_ac3_pay_finalize (GObject * object); static GstStateChangeReturn gst_rtp_ac3_pay_change_state (GstElement * element, GstStateChange transition); -static gboolean gst_rtp_ac3_pay_setcaps (GstBaseRTPPayload * payload, +static gboolean gst_rtp_ac3_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps); -static gboolean gst_rtp_ac3_pay_handle_event (GstBaseRTPPayload * payload, +static gboolean gst_rtp_ac3_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event); static GstFlowReturn gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay); -static GstFlowReturn gst_rtp_ac3_pay_handle_buffer (GstBaseRTPPayload * payload, +static GstFlowReturn gst_rtp_ac3_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer); #define gst_rtp_ac3_pay_parent_class parent_class -G_DEFINE_TYPE (GstRtpAC3Pay, gst_rtp_ac3_pay, GST_TYPE_BASE_RTP_PAYLOAD); +G_DEFINE_TYPE (GstRtpAC3Pay, gst_rtp_ac3_pay, GST_TYPE_RTP_BASE_PAYLOAD); static void gst_rtp_ac3_pay_class_init (GstRtpAC3PayClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; - GstBaseRTPPayloadClass *gstbasertppayload_class; + GstRTPBasePayloadClass *gstrtpbasepayload_class; GST_DEBUG_CATEGORY_INIT (rtpac3pay_debug, "rtpac3pay", 0, "AC3 Audio RTP Depayloader"); gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; - gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; + gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass; gobject_class->finalize = gst_rtp_ac3_pay_finalize; gstelement_class->change_state = gst_rtp_ac3_pay_change_state; - gst_element_class_add_pad_template (gstelement_class, - gst_static_pad_template_get (&gst_rtp_ac3_pay_src_template)); - gst_element_class_add_pad_template (gstelement_class, - gst_static_pad_template_get (&gst_rtp_ac3_pay_sink_template)); + gst_element_class_add_static_pad_template (gstelement_class, + &gst_rtp_ac3_pay_src_template); + gst_element_class_add_static_pad_template (gstelement_class, + &gst_rtp_ac3_pay_sink_template); - gst_element_class_set_details_simple (gstelement_class, + gst_element_class_set_static_metadata (gstelement_class, "RTP AC3 audio payloader", "Codec/Payloader/Network/RTP", "Payload AC3 audio as RTP packets (RFC 4184)", "Wim Taymans "); - gstbasertppayload_class->set_caps = gst_rtp_ac3_pay_setcaps; - gstbasertppayload_class->handle_event = gst_rtp_ac3_pay_handle_event; - gstbasertppayload_class->handle_buffer = gst_rtp_ac3_pay_handle_buffer; + gstrtpbasepayload_class->set_caps = gst_rtp_ac3_pay_setcaps; + gstrtpbasepayload_class->sink_event = gst_rtp_ac3_pay_sink_event; + gstrtpbasepayload_class->handle_buffer = gst_rtp_ac3_pay_handle_buffer; } static void @@ -125,7 +143,7 @@ gst_rtp_ac3_pay_reset (GstRtpAC3Pay * pay) } static gboolean -gst_rtp_ac3_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) +gst_rtp_ac3_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps) { gboolean res; gint rate; @@ -136,14 +154,14 @@ gst_rtp_ac3_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) if (!gst_structure_get_int (structure, "rate", &rate)) rate = 90000; /* default */ - gst_basertppayload_set_options (payload, "audio", TRUE, "AC3", rate); - res = gst_basertppayload_set_outcaps (payload, NULL); + gst_rtp_base_payload_set_options (payload, "audio", TRUE, "AC3", rate); + res = gst_rtp_base_payload_set_outcaps (payload, NULL); return res; } static gboolean -gst_rtp_ac3_pay_handle_event (GstBaseRTPPayload * payload, GstEvent * event) +gst_rtp_ac3_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event) { gboolean res; GstRtpAC3Pay *rtpac3pay; @@ -162,8 +180,7 @@ gst_rtp_ac3_pay_handle_event (GstBaseRTPPayload * payload, GstEvent * event) break; } - res = - GST_BASE_RTP_PAYLOAD_CLASS (parent_class)->handle_event (payload, event); + res = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event); return res; } @@ -235,7 +252,7 @@ gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay) /* number of frames */ NF = rtpac3pay->NF; - mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpac3pay); + mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpac3pay); GST_LOG_OBJECT (rtpac3pay, "flushing %u bytes", avail); @@ -245,6 +262,7 @@ gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay) guint payload_len; guint packet_len; GstRTPBuffer rtp = { NULL, }; + GstBuffer *payload_buffer; /* this will be the total length of the packet */ packet_len = gst_rtp_buffer_calc_packet_len (2 + avail, 0, 0); @@ -256,7 +274,9 @@ gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay) payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0); /* create buffer to hold the payload */ - outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); + outbuf = + gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD + (rtpac3pay), 2, 0, 0); if (FT == 0) { /* check if it all fits */ @@ -299,49 +319,55 @@ gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay) payload[1] = NF; payload_len -= 2; - gst_adapter_copy (rtpac3pay->adapter, &payload[2], 0, payload_len); - gst_adapter_flush (rtpac3pay->adapter, payload_len); - - avail -= payload_len; - if (avail == 0) + if (avail == payload_len) gst_rtp_buffer_set_marker (&rtp, TRUE); gst_rtp_buffer_unmap (&rtp); - GST_BUFFER_TIMESTAMP (outbuf) = rtpac3pay->first_ts; + payload_buffer = + gst_adapter_take_buffer_fast (rtpac3pay->adapter, payload_len); + + gst_rtp_copy_audio_meta (rtpac3pay, outbuf, payload_buffer); + + outbuf = gst_buffer_append (outbuf, payload_buffer); + + avail -= payload_len; + + GST_BUFFER_PTS (outbuf) = rtpac3pay->first_ts; GST_BUFFER_DURATION (outbuf) = rtpac3pay->duration; - ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpac3pay), outbuf); + ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpac3pay), outbuf); } return ret; } static GstFlowReturn -gst_rtp_ac3_pay_handle_buffer (GstBaseRTPPayload * basepayload, +gst_rtp_ac3_pay_handle_buffer (GstRTPBasePayload * basepayload, GstBuffer * buffer) { GstRtpAC3Pay *rtpac3pay; GstFlowReturn ret; - gsize size, avail, left, NF; - guint8 *data, *p; + gsize avail, left, NF; + GstMapInfo map; + guint8 *p; guint packet_len; GstClockTime duration, timestamp; rtpac3pay = GST_RTP_AC3_PAY (basepayload); - data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ); + gst_buffer_map (buffer, &map, GST_MAP_READ); duration = GST_BUFFER_DURATION (buffer); - timestamp = GST_BUFFER_TIMESTAMP (buffer); + timestamp = GST_BUFFER_PTS (buffer); if (GST_BUFFER_IS_DISCONT (buffer)) { GST_DEBUG_OBJECT (rtpac3pay, "DISCONT"); gst_rtp_ac3_pay_reset (rtpac3pay); } - /* count the amount of incomming packets */ + /* count the amount of incoming packets */ NF = 0; - left = size; - p = data; + left = map.size; + p = map.data; while (TRUE) { guint bsid, fscod, frmsizecod, frame_size; @@ -368,12 +394,13 @@ gst_rtp_ac3_pay_handle_buffer (GstBaseRTPPayload * basepayload, break; NF++; - GST_DEBUG_OBJECT (rtpac3pay, "found frame %u of size %u", NF, frame_size); + GST_DEBUG_OBJECT (rtpac3pay, "found frame %" G_GSIZE_FORMAT " of size %u", + NF, frame_size); p += frame_size; left -= frame_size; } - gst_buffer_unmap (buffer, data, size); + gst_buffer_unmap (buffer, &map); if (NF == 0) goto no_frames; @@ -381,11 +408,11 @@ gst_rtp_ac3_pay_handle_buffer (GstBaseRTPPayload * basepayload, /* get packet length of previous data and this new data, * payload length includes a 4 byte header */ - packet_len = gst_rtp_buffer_calc_packet_len (2 + avail + size, 0, 0); + packet_len = gst_rtp_buffer_calc_packet_len (2 + avail + map.size, 0, 0); /* if this buffer is going to overflow the packet, flush what we * have. */ - if (gst_basertppayload_is_filled (basepayload, + if (gst_rtp_base_payload_is_filled (basepayload, packet_len, rtpac3pay->duration + duration)) { ret = gst_rtp_ac3_pay_flush (rtpac3pay); avail = 0;