--- /dev/null
+/* GStreamer
+ * Copyright (C) <2010> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include <string.h>
+
+#include <gst/rtp/gstrtpbuffer.h>
+
+#include "gstrtpac3pay.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpac3pay_debug);
+#define GST_CAT_DEFAULT (rtpac3pay_debug)
+
+static GstStaticPadTemplate gst_rtp_ac3_pay_sink_template =
+ GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/ac3; " "audio/x-ac3; ")
+ );
+
+static GstStaticPadTemplate gst_rtp_ac3_pay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtp, "
+ "media = (string) \"audio\", "
+ "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+ "clock-rate = (int) { 32000, 44100, 48000 }, "
+ "encoding-name = (string) \"AC3\"")
+ );
+
+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,
+ GstCaps * caps);
+static gboolean gst_rtp_ac3_pay_handle_event (GstPad * pad, GstEvent * event);
+static GstFlowReturn gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay);
+static GstFlowReturn gst_rtp_ac3_pay_handle_buffer (GstBaseRTPPayload * payload,
+ GstBuffer * buffer);
+
+GST_BOILERPLATE (GstRtpAC3Pay, gst_rtp_ac3_pay, GstBaseRTPPayload,
+ GST_TYPE_BASE_RTP_PAYLOAD)
+
+ static void gst_rtp_ac3_pay_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtp_ac3_pay_src_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtp_ac3_pay_sink_template));
+
+ gst_element_class_set_details_simple (element_class,
+ "RTP AC3 audio payloader", "Codec/Payloader/Network/RTP",
+ "Payload AC3 audio as RTP packets (RFC 4184)",
+ "Wim Taymans <wim.taymans@gmail.com>");
+}
+
+static void
+gst_rtp_ac3_pay_class_init (GstRtpAC3PayClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+
+ gobject_class->finalize = gst_rtp_ac3_pay_finalize;
+
+ gstelement_class->change_state = gst_rtp_ac3_pay_change_state;
+
+ 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;
+
+ GST_DEBUG_CATEGORY_INIT (rtpac3pay_debug, "rtpac3pay", 0,
+ "AC3 Audio RTP Depayloader");
+}
+
+static void
+gst_rtp_ac3_pay_init (GstRtpAC3Pay * rtpac3pay, GstRtpAC3PayClass * klass)
+{
+ rtpac3pay->adapter = gst_adapter_new ();
+}
+
+static void
+gst_rtp_ac3_pay_finalize (GObject * object)
+{
+ GstRtpAC3Pay *rtpac3pay;
+
+ rtpac3pay = GST_RTP_AC3_PAY (object);
+
+ g_object_unref (rtpac3pay->adapter);
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static void
+gst_rtp_ac3_pay_reset (GstRtpAC3Pay * pay)
+{
+ pay->first_ts = -1;
+ pay->duration = 0;
+ gst_adapter_clear (pay->adapter);
+ GST_DEBUG_OBJECT (pay, "reset depayloader");
+}
+
+static gboolean
+gst_rtp_ac3_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
+{
+ gboolean res;
+ gint rate;
+ GstStructure *structure;
+
+ structure = gst_caps_get_structure (caps, 0);
+
+ 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);
+
+ return res;
+}
+
+static gboolean
+gst_rtp_ac3_pay_handle_event (GstPad * pad, GstEvent * event)
+{
+ GstRtpAC3Pay *rtpac3pay;
+
+ rtpac3pay = GST_RTP_AC3_PAY (gst_pad_get_parent (pad));
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_EOS:
+ /* make sure we push the last packets in the adapter on EOS */
+ gst_rtp_ac3_pay_flush (rtpac3pay);
+ break;
+ case GST_EVENT_FLUSH_STOP:
+ gst_rtp_ac3_pay_reset (rtpac3pay);
+ break;
+ default:
+ break;
+ }
+
+ gst_object_unref (rtpac3pay);
+
+ /* FALSE to let the parent handle the event as well */
+ return FALSE;
+}
+
+struct frmsize_s
+{
+ guint16 bit_rate;
+ guint16 frm_size[3];
+};
+
+static const struct frmsize_s frmsizecod_tbl[] = {
+ {32, {64, 69, 96}},
+ {32, {64, 70, 96}},
+ {40, {80, 87, 120}},
+ {40, {80, 88, 120}},
+ {48, {96, 104, 144}},
+ {48, {96, 105, 144}},
+ {56, {112, 121, 168}},
+ {56, {112, 122, 168}},
+ {64, {128, 139, 192}},
+ {64, {128, 140, 192}},
+ {80, {160, 174, 240}},
+ {80, {160, 175, 240}},
+ {96, {192, 208, 288}},
+ {96, {192, 209, 288}},
+ {112, {224, 243, 336}},
+ {112, {224, 244, 336}},
+ {128, {256, 278, 384}},
+ {128, {256, 279, 384}},
+ {160, {320, 348, 480}},
+ {160, {320, 349, 480}},
+ {192, {384, 417, 576}},
+ {192, {384, 418, 576}},
+ {224, {448, 487, 672}},
+ {224, {448, 488, 672}},
+ {256, {512, 557, 768}},
+ {256, {512, 558, 768}},
+ {320, {640, 696, 960}},
+ {320, {640, 697, 960}},
+ {384, {768, 835, 1152}},
+ {384, {768, 836, 1152}},
+ {448, {896, 975, 1344}},
+ {448, {896, 976, 1344}},
+ {512, {1024, 1114, 1536}},
+ {512, {1024, 1115, 1536}},
+ {576, {1152, 1253, 1728}},
+ {576, {1152, 1254, 1728}},
+ {640, {1280, 1393, 1920}},
+ {640, {1280, 1394, 1920}}
+};
+
+static GstFlowReturn
+gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay)
+{
+ guint avail, FT, NF, mtu;
+ GstBuffer *outbuf;
+ GstFlowReturn ret;
+
+ /* the data available in the adapter is either smaller
+ * than the MTU or bigger. In the case it is smaller, the complete
+ * adapter contents can be put in one packet. In the case the
+ * adapter has more than one MTU, we need to split the AC3 data
+ * over multiple packets. */
+ avail = gst_adapter_available (rtpac3pay->adapter);
+
+ ret = GST_FLOW_OK;
+
+ FT = 0;
+ /* number of frames */
+ NF = rtpac3pay->NF;
+
+ mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpac3pay);
+
+ GST_LOG_OBJECT (rtpac3pay, "flushing %u bytes", avail);
+
+ while (avail > 0) {
+ guint towrite;
+ guint8 *payload;
+ guint payload_len;
+ guint packet_len;
+
+ /* this will be the total length of the packet */
+ packet_len = gst_rtp_buffer_calc_packet_len (2 + avail, 0, 0);
+
+ /* fill one MTU or all available bytes */
+ towrite = MIN (packet_len, mtu);
+
+ /* this is the payload length */
+ 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);
+
+ if (FT == 0) {
+ /* check if it all fits */
+ if (towrite < packet_len) {
+ guint maxlen;
+
+ GST_LOG_OBJECT (rtpac3pay, "we need to fragment");
+ /* check if we will be able to put at least 5/8th of the total
+ * frame in this first frame. */
+ if ((avail * 5) / 8 >= (payload_len - 2))
+ FT = 1;
+ else
+ FT = 2;
+ /* check how many fragments we will need */
+ maxlen = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0);
+ NF = (avail + maxlen - 1) / maxlen;
+ }
+ } else if (FT != 3) {
+ /* remaining fragment */
+ FT = 3;
+ }
+
+ /*
+ * 0 1
+ * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+ * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ * | MBZ | FT| NF |
+ * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ *
+ * FT: 0: one or more complete frames
+ * 1: initial 5/8 fragment
+ * 2: initial fragment not 5/8
+ * 3: other fragment
+ * NF: amount of frames if FT = 0, else number of fragments.
+ */
+ GST_LOG_OBJECT (rtpac3pay, "FT %u, NF %u", FT, NF);
+ payload = gst_rtp_buffer_get_payload (outbuf);
+ payload[0] = (FT & 3);
+ 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)
+ gst_rtp_buffer_set_marker (outbuf, TRUE);
+
+ GST_BUFFER_TIMESTAMP (outbuf) = rtpac3pay->first_ts;
+ GST_BUFFER_DURATION (outbuf) = rtpac3pay->duration;
+
+ ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpac3pay), outbuf);
+ }
+
+ return ret;
+}
+
+static GstFlowReturn
+gst_rtp_ac3_pay_handle_buffer (GstBaseRTPPayload * basepayload,
+ GstBuffer * buffer)
+{
+ GstRtpAC3Pay *rtpac3pay;
+ GstFlowReturn ret;
+ guint size, avail, left, NF;
+ guint8 *data, *p;
+ guint packet_len;
+ GstClockTime duration, timestamp;
+
+ rtpac3pay = GST_RTP_AC3_PAY (basepayload);
+
+ size = GST_BUFFER_SIZE (buffer);
+ data = GST_BUFFER_DATA (buffer);
+ duration = GST_BUFFER_DURATION (buffer);
+ timestamp = GST_BUFFER_TIMESTAMP (buffer);
+
+ if (GST_BUFFER_IS_DISCONT (buffer)) {
+ GST_DEBUG_OBJECT (rtpac3pay, "DISCONT");
+ gst_rtp_ac3_pay_reset (rtpac3pay);
+ }
+
+ /* count the amount of incomming packets */
+ NF = 0;
+ left = size;
+ p = data;
+ while (TRUE) {
+ guint bsid, fscod, frmsizecod, frame_size;
+
+ if (left < 6)
+ break;
+
+ if (p[0] != 0x0b || p[1] != 0x77)
+ break;
+
+ bsid = p[5] >> 3;
+ if (bsid > 8)
+ break;
+
+ frmsizecod = p[4] & 0x3f;
+ fscod = p[4] >> 6;
+
+ GST_DEBUG_OBJECT (rtpac3pay, "fscod %u, %u", fscod, frmsizecod);
+
+ if (fscod >= 3 || frmsizecod >= 38)
+ break;
+
+ frame_size = frmsizecod_tbl[frmsizecod].frm_size[fscod] * 2;
+ if (frame_size > left)
+ break;
+
+ NF++;
+ GST_DEBUG_OBJECT (rtpac3pay, "found frame %u of size %u", NF, frame_size);
+
+ p += frame_size;
+ left -= frame_size;
+ }
+ if (NF == 0)
+ goto no_frames;
+
+ avail = gst_adapter_available (rtpac3pay->adapter);
+
+ /* 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);
+
+ /* if this buffer is going to overflow the packet, flush what we
+ * have. */
+ if (gst_basertppayload_is_filled (basepayload,
+ packet_len, rtpac3pay->duration + duration)) {
+ ret = gst_rtp_ac3_pay_flush (rtpac3pay);
+ avail = 0;
+ } else {
+ ret = GST_FLOW_OK;
+ }
+
+ if (avail == 0) {
+ GST_DEBUG_OBJECT (rtpac3pay,
+ "first packet, save timestamp %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (timestamp));
+ rtpac3pay->first_ts = timestamp;
+ rtpac3pay->duration = 0;
+ rtpac3pay->NF = 0;
+ }
+
+ gst_adapter_push (rtpac3pay->adapter, buffer);
+ rtpac3pay->duration += duration;
+ rtpac3pay->NF += NF;
+
+ return ret;
+
+ /* ERRORS */
+no_frames:
+ {
+ GST_WARNING_OBJECT (rtpac3pay, "no valid AC3 frames found");
+ return GST_FLOW_OK;
+ }
+}
+
+static GstStateChangeReturn
+gst_rtp_ac3_pay_change_state (GstElement * element, GstStateChange transition)
+{
+ GstRtpAC3Pay *rtpac3pay;
+ GstStateChangeReturn ret;
+
+ rtpac3pay = GST_RTP_AC3_PAY (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ gst_rtp_ac3_pay_reset (rtpac3pay);
+ break;
+ default:
+ break;
+ }
+
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ gst_rtp_ac3_pay_reset (rtpac3pay);
+ break;
+ default:
+ break;
+ }
+ return ret;
+}
+
+gboolean
+gst_rtp_ac3_pay_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpac3pay",
+ GST_RANK_SECONDARY, GST_TYPE_RTP_AC3_PAY);
+}
projects / platform / upstream / gst-plugins-good.git / commitdiff