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/* GStreamer
* Copyright (C) <2005> 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.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/video/video.h>
#include "gstrtpmp4vpay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpmp4vpay_debug);
#define GST_CAT_DEFAULT (rtpmp4vpay_debug)
static GstStaticPadTemplate gst_rtp_mp4v_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/mpeg,"
"mpegversion=(int) 4, systemstream=(boolean)false;" "video/x-divx")
);
static GstStaticPadTemplate gst_rtp_mp4v_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"video\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"MP4V-ES\""
/* two string params
*
"profile-level-id = (string) [1,MAX]"
"config = (string) [1,MAX]"
*/
)
);
#define DEFAULT_CONFIG_INTERVAL 0
enum
{
PROP_0,
PROP_CONFIG_INTERVAL
};
static void gst_rtp_mp4v_pay_finalize (GObject * object);
static void gst_rtp_mp4v_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_mp4v_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_rtp_mp4v_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_mp4v_pay_handle_buffer (GstRTPBasePayload *
payload, GstBuffer * buffer);
static gboolean gst_rtp_mp4v_pay_sink_event (GstRTPBasePayload * pay,
GstEvent * event);
#define gst_rtp_mp4v_pay_parent_class parent_class
G_DEFINE_TYPE (GstRtpMP4VPay, gst_rtp_mp4v_pay, GST_TYPE_RTP_BASE_PAYLOAD)
static void gst_rtp_mp4v_pay_class_init (GstRtpMP4VPayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gobject_class->set_property = gst_rtp_mp4v_pay_set_property;
gobject_class->get_property = gst_rtp_mp4v_pay_get_property;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_mp4v_pay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_mp4v_pay_sink_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP MPEG4 Video payloader", "Codec/Payloader/Network/RTP",
"Payload MPEG-4 video as RTP packets (RFC 3016)",
"Wim Taymans <wim.taymans@gmail.com>");
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CONFIG_INTERVAL,
g_param_spec_uint ("config-interval", "Config Send Interval",
"Send Config Insertion Interval in seconds (configuration headers "
"will be multiplexed in the data stream when detected.) (0 = disabled)",
0, 3600, DEFAULT_CONFIG_INTERVAL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
);
gobject_class->finalize = gst_rtp_mp4v_pay_finalize;
gstrtpbasepayload_class->set_caps = gst_rtp_mp4v_pay_setcaps;
gstrtpbasepayload_class->handle_buffer = gst_rtp_mp4v_pay_handle_buffer;
gstrtpbasepayload_class->sink_event = gst_rtp_mp4v_pay_sink_event;
GST_DEBUG_CATEGORY_INIT (rtpmp4vpay_debug, "rtpmp4vpay", 0,
"MP4 video RTP Payloader");
}
static void
gst_rtp_mp4v_pay_init (GstRtpMP4VPay * rtpmp4vpay)
{
rtpmp4vpay->adapter = gst_adapter_new ();
rtpmp4vpay->rate = 90000;
rtpmp4vpay->profile = 1;
rtpmp4vpay->need_config = TRUE;
rtpmp4vpay->config_interval = DEFAULT_CONFIG_INTERVAL;
rtpmp4vpay->last_config = -1;
rtpmp4vpay->config = NULL;
}
static void
gst_rtp_mp4v_pay_finalize (GObject * object)
{
GstRtpMP4VPay *rtpmp4vpay;
rtpmp4vpay = GST_RTP_MP4V_PAY (object);
if (rtpmp4vpay->config) {
gst_buffer_unref (rtpmp4vpay->config);
rtpmp4vpay->config = NULL;
}
g_object_unref (rtpmp4vpay->adapter);
rtpmp4vpay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_mp4v_pay_new_caps (GstRtpMP4VPay * rtpmp4vpay)
{
gchar *profile, *config;
GValue v = { 0 };
gboolean res;
profile = g_strdup_printf ("%d", rtpmp4vpay->profile);
g_value_init (&v, GST_TYPE_BUFFER);
gst_value_set_buffer (&v, rtpmp4vpay->config);
config = gst_value_serialize (&v);
res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpmp4vpay),
"profile-level-id", G_TYPE_STRING, profile,
"config", G_TYPE_STRING, config, NULL);
g_value_unset (&v);
g_free (profile);
g_free (config);
return res;
}
static gboolean
gst_rtp_mp4v_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
GstRtpMP4VPay *rtpmp4vpay;
GstStructure *structure;
const GValue *codec_data;
gboolean res;
rtpmp4vpay = GST_RTP_MP4V_PAY (payload);
gst_rtp_base_payload_set_options (payload, "video", TRUE, "MP4V-ES",
rtpmp4vpay->rate);
res = TRUE;
structure = gst_caps_get_structure (caps, 0);
codec_data = gst_structure_get_value (structure, "codec_data");
if (codec_data) {
GST_LOG_OBJECT (rtpmp4vpay, "got codec_data");
if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) {
GstBuffer *buffer;
buffer = gst_value_get_buffer (codec_data);
if (gst_buffer_get_size (buffer) < 5)
goto done;
gst_buffer_extract (buffer, 4, &rtpmp4vpay->profile, 1);
GST_LOG_OBJECT (rtpmp4vpay, "configuring codec_data, profile %d",
rtpmp4vpay->profile);
if (rtpmp4vpay->config)
gst_buffer_unref (rtpmp4vpay->config);
rtpmp4vpay->config = gst_buffer_copy (buffer);
res = gst_rtp_mp4v_pay_new_caps (rtpmp4vpay);
}
}
done:
return res;
}
static void
gst_rtp_mp4v_pay_empty (GstRtpMP4VPay * rtpmp4vpay)
{
gst_adapter_clear (rtpmp4vpay->adapter);
}
#define RTP_HEADER_LEN 12
static GstFlowReturn
gst_rtp_mp4v_pay_flush (GstRtpMP4VPay * rtpmp4vpay)
{
guint avail, mtu;
GstBuffer *outbuf;
GstBuffer *outbuf_data = NULL;
GstFlowReturn ret;
GstBufferList *list = NULL;
/* 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 MP4V data
* over multiple packets. */
avail = gst_adapter_available (rtpmp4vpay->adapter);
if (rtpmp4vpay->config == NULL && rtpmp4vpay->need_config) {
/* when we don't have a config yet, flush things out */
gst_adapter_flush (rtpmp4vpay->adapter, avail);
avail = 0;
}
if (!avail)
return GST_FLOW_OK;
mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp4vpay);
/* Use buffer lists. Each frame will be put into a list
* of buffers and the whole list will be pushed downstream
* at once */
list = gst_buffer_list_new_sized ((avail / (mtu - RTP_HEADER_LEN)) + 1);
while (avail > 0) {
guint towrite;
guint payload_len;
guint packet_len;
GstRTPBuffer rtp = { NULL };
/* this will be the total lenght of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (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 without payload. The payload will be put
* in next buffer instead. Both buffers will be merged */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
/* Take buffer with the payload from the adapter */
outbuf_data = gst_adapter_take_buffer_fast (rtpmp4vpay->adapter,
payload_len);
avail -= payload_len;
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
gst_rtp_buffer_set_marker (&rtp, avail == 0);
gst_rtp_buffer_unmap (&rtp);
gst_rtp_copy_video_meta (rtpmp4vpay, outbuf, outbuf_data);
outbuf = gst_buffer_append (outbuf, outbuf_data);
GST_BUFFER_PTS (outbuf) = rtpmp4vpay->first_timestamp;
/* add to list */
gst_buffer_list_insert (list, -1, outbuf);
}
/* push the whole buffer list at once */
ret =
gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtpmp4vpay), list);
return ret;
}
#define VOS_STARTCODE 0x000001B0
#define VOS_ENDCODE 0x000001B1
#define USER_DATA_STARTCODE 0x000001B2
#define GOP_STARTCODE 0x000001B3
#define VISUAL_OBJECT_STARTCODE 0x000001B5
#define VOP_STARTCODE 0x000001B6
static gboolean
gst_rtp_mp4v_pay_depay_data (GstRtpMP4VPay * enc, guint8 * data, guint size,
gint * strip, gboolean * vopi)
{
guint32 code;
gboolean result;
*vopi = FALSE;
*strip = 0;
if (size < 5)
return FALSE;
code = GST_READ_UINT32_BE (data);
GST_DEBUG_OBJECT (enc, "start code 0x%08x", code);
switch (code) {
case VOS_STARTCODE:
case 0x00000101:
{
gint i;
guint8 profile;
gboolean newprofile = FALSE;
gboolean equal;
if (code == VOS_STARTCODE) {
/* profile_and_level_indication */
profile = data[4];
GST_DEBUG_OBJECT (enc, "VOS profile 0x%08x", profile);
if (profile != enc->profile) {
newprofile = TRUE;
enc->profile = profile;
}
}
/* up to the next GOP_STARTCODE or VOP_STARTCODE is
* the config information */
code = 0xffffffff;
for (i = 5; i < size - 4; i++) {
code = (code << 8) | data[i];
if (code == GOP_STARTCODE || code == VOP_STARTCODE)
break;
}
i -= 3;
/* see if config changed */
equal = FALSE;
if (enc->config) {
if (gst_buffer_get_size (enc->config) == i) {
equal = gst_buffer_memcmp (enc->config, 0, data, i) == 0;
}
}
/* if config string changed or new profile, make new caps */
if (!equal || newprofile) {
if (enc->config)
gst_buffer_unref (enc->config);
enc->config = gst_buffer_new_and_alloc (i);
gst_buffer_fill (enc->config, 0, data, i);
gst_rtp_mp4v_pay_new_caps (enc);
}
*strip = i;
/* we need to flush out the current packet. */
result = TRUE;
break;
}
case VOP_STARTCODE:
GST_DEBUG_OBJECT (enc, "VOP");
/* VOP startcode, we don't have to flush the packet */
result = FALSE;
/* vop-coding-type == I-frame */
if (size > 4 && (data[4] >> 6 == 0)) {
GST_DEBUG_OBJECT (enc, "VOP-I");
*vopi = TRUE;
}
break;
case GOP_STARTCODE:
GST_DEBUG_OBJECT (enc, "GOP");
*vopi = TRUE;
result = TRUE;
break;
case 0x00000100:
enc->need_config = FALSE;
result = TRUE;
break;
default:
if (code >= 0x20 && code <= 0x2f) {
GST_DEBUG_OBJECT (enc, "short header");
result = FALSE;
} else {
GST_DEBUG_OBJECT (enc, "other startcode");
/* all other startcodes need a flush */
result = TRUE;
}
break;
}
return result;
}
/* we expect buffers starting on startcodes.
*/
static GstFlowReturn
gst_rtp_mp4v_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpMP4VPay *rtpmp4vpay;
GstFlowReturn ret;
guint avail;
guint packet_len;
GstMapInfo map;
gsize size;
gboolean flush;
gint strip;
GstClockTime timestamp, duration;
gboolean vopi;
gboolean send_config;
ret = GST_FLOW_OK;
send_config = FALSE;
rtpmp4vpay = GST_RTP_MP4V_PAY (basepayload);
gst_buffer_map (buffer, &map, GST_MAP_READ);
size = map.size;
timestamp = GST_BUFFER_PTS (buffer);
duration = GST_BUFFER_DURATION (buffer);
avail = gst_adapter_available (rtpmp4vpay->adapter);
if (duration == -1)
duration = 0;
/* empty buffer, take timestamp */
if (avail == 0) {
rtpmp4vpay->first_timestamp = timestamp;
rtpmp4vpay->duration = 0;
}
/* depay incomming data and see if we need to start a new RTP
* packet */
flush =
gst_rtp_mp4v_pay_depay_data (rtpmp4vpay, map.data, size, &strip, &vopi);
gst_buffer_unmap (buffer, &map);
if (strip) {
/* strip off config if requested */
if (!(rtpmp4vpay->config_interval > 0)) {
GstBuffer *subbuf;
GST_LOG_OBJECT (rtpmp4vpay, "stripping config at %d, size %d", strip,
(gint) size - strip);
/* strip off header */
subbuf = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, strip,
size - strip);
GST_BUFFER_PTS (subbuf) = timestamp;
gst_buffer_unref (buffer);
buffer = subbuf;
size = gst_buffer_get_size (buffer);
} else {
GST_LOG_OBJECT (rtpmp4vpay, "found config in stream");
rtpmp4vpay->last_config = timestamp;
}
}
/* there is a config request, see if we need to insert it */
if (vopi && (rtpmp4vpay->config_interval > 0) && rtpmp4vpay->config) {
if (rtpmp4vpay->last_config != -1) {
guint64 diff;
GST_LOG_OBJECT (rtpmp4vpay,
"now %" GST_TIME_FORMAT ", last VOP-I %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp), GST_TIME_ARGS (rtpmp4vpay->last_config));
/* calculate diff between last config in milliseconds */
if (timestamp > rtpmp4vpay->last_config) {
diff = timestamp - rtpmp4vpay->last_config;
} else {
diff = 0;
}
GST_DEBUG_OBJECT (rtpmp4vpay,
"interval since last config %" GST_TIME_FORMAT, GST_TIME_ARGS (diff));
/* bigger than interval, queue config */
/* FIXME should convert timestamps to running time */
if (GST_TIME_AS_SECONDS (diff) >= rtpmp4vpay->config_interval) {
GST_DEBUG_OBJECT (rtpmp4vpay, "time to send config");
send_config = TRUE;
}
} else {
/* no known previous config time, send now */
GST_DEBUG_OBJECT (rtpmp4vpay, "no previous config time, send now");
send_config = TRUE;
}
if (send_config) {
/* we need to send config now first */
GST_LOG_OBJECT (rtpmp4vpay, "inserting config in stream");
/* insert header */
buffer = gst_buffer_append (gst_buffer_ref (rtpmp4vpay->config), buffer);
GST_BUFFER_PTS (buffer) = timestamp;
size = gst_buffer_get_size (buffer);
if (timestamp != -1) {
rtpmp4vpay->last_config = timestamp;
}
}
}
/* if we need to flush, do so now */
if (flush) {
ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay);
rtpmp4vpay->first_timestamp = timestamp;
rtpmp4vpay->duration = 0;
avail = 0;
}
/* get packet length of data and see if we exceeded MTU. */
packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
if (gst_rtp_base_payload_is_filled (basepayload,
packet_len, rtpmp4vpay->duration + duration)) {
ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay);
rtpmp4vpay->first_timestamp = timestamp;
rtpmp4vpay->duration = 0;
}
/* push new data */
gst_adapter_push (rtpmp4vpay->adapter, buffer);
rtpmp4vpay->duration += duration;
return ret;
}
static gboolean
gst_rtp_mp4v_pay_sink_event (GstRTPBasePayload * pay, GstEvent * event)
{
GstRtpMP4VPay *rtpmp4vpay;
rtpmp4vpay = GST_RTP_MP4V_PAY (pay);
GST_DEBUG ("Got event: %s", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:
case GST_EVENT_EOS:
/* This flush call makes sure that the last buffer is always pushed
* to the base payloader */
gst_rtp_mp4v_pay_flush (rtpmp4vpay);
break;
case GST_EVENT_FLUSH_STOP:
gst_rtp_mp4v_pay_empty (rtpmp4vpay);
break;
default:
break;
}
/* let parent handle event too */
return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (pay, event);
}
static void
gst_rtp_mp4v_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpMP4VPay *rtpmp4vpay;
rtpmp4vpay = GST_RTP_MP4V_PAY (object);
switch (prop_id) {
case PROP_CONFIG_INTERVAL:
rtpmp4vpay->config_interval = g_value_get_uint (value);
break;
default:
break;
}
}
static void
gst_rtp_mp4v_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpMP4VPay *rtpmp4vpay;
rtpmp4vpay = GST_RTP_MP4V_PAY (object);
switch (prop_id) {
case PROP_CONFIG_INTERVAL:
g_value_set_uint (value, rtpmp4vpay->config_interval);
break;
default:
break;
}
}
gboolean
gst_rtp_mp4v_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpmp4vpay",
GST_RANK_SECONDARY, GST_TYPE_RTP_MP4V_PAY);
}