/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-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
*
* <refsect2>
* <title>Example pipeline</title>
* |[
* 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.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "gstrtpac3pay.h"
#include "gstrtputils.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 (GstRTPBasePayload * payload,
GstCaps * caps);
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 (GstRTPBasePayload * payload,
GstBuffer * buffer);
#define gst_rtp_ac3_pay_parent_class parent_class
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;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
GST_DEBUG_CATEGORY_INIT (rtpac3pay_debug, "rtpac3pay", 0,
"AC3 Audio RTP Depayloader");
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) 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_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_static_metadata (gstelement_class,
"RTP AC3 audio payloader", "Codec/Payloader/Network/RTP",
"Payload AC3 audio as RTP packets (RFC 4184)",
"Wim Taymans <wim.taymans@gmail.com>");
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
gst_rtp_ac3_pay_init (GstRtpAC3Pay * rtpac3pay)
{
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 (GstRTPBasePayload * 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_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_sink_event (GstRTPBasePayload * payload, GstEvent * event)
{
gboolean res;
GstRtpAC3Pay *rtpac3pay;
rtpac3pay = GST_RTP_AC3_PAY (payload);
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;
}
res = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
return res;
}
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_RTP_BASE_PAYLOAD_MTU (rtpac3pay);
GST_LOG_OBJECT (rtpac3pay, "flushing %u bytes", avail);
while (avail > 0) {
guint towrite;
guint8 *payload;
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);
/* 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 (2, 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_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
GST_LOG_OBJECT (rtpac3pay, "FT %u, NF %u", FT, NF);
payload = gst_rtp_buffer_get_payload (&rtp);
payload[0] = (FT & 3);
payload[1] = NF;
payload_len -= 2;
if (avail == payload_len)
gst_rtp_buffer_set_marker (&rtp, TRUE);
gst_rtp_buffer_unmap (&rtp);
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_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpac3pay), outbuf);
}
return ret;
}
static GstFlowReturn
gst_rtp_ac3_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpAC3Pay *rtpac3pay;
GstFlowReturn ret;
gsize avail, left, NF;
GstMapInfo map;
guint8 *p;
guint packet_len;
GstClockTime duration, timestamp;
rtpac3pay = GST_RTP_AC3_PAY (basepayload);
gst_buffer_map (buffer, &map, GST_MAP_READ);
duration = GST_BUFFER_DURATION (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 incoming packets */
NF = 0;
left = map.size;
p = map.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 %" G_GSIZE_FORMAT " of size %u",
NF, frame_size);
p += frame_size;
left -= frame_size;
}
gst_buffer_unmap (buffer, &map);
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 + map.size, 0, 0);
/* if this buffer is going to overflow the packet, flush what we
* have. */
if (gst_rtp_base_payload_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);
}