gstreamer/gst/rtp/gstrtpmp4vpay.c
Sjoerd Simons f34fce9df4 gst/rtp/gstrtpmp4vpay.*: Handle NEWSEGMENT and FLUSH events. Fixes #434824.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init),
(gst_rtp_mp4v_pay_empty), (gst_rtp_mp4v_pay_event):
* gst/rtp/gstrtpmp4vpay.h:
Handle NEWSEGMENT and FLUSH events. Fixes #434824.
2007-05-01 16:13:58 +00:00

536 lines
14 KiB
C

/* GStreamer
* Copyright (C) <2005> Wim Taymans <wim@fluendo.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 "gstrtpmp4vpay.h"
GST_DEBUG_CATEGORY_STATIC (rtpmp4vpay_debug);
#define GST_CAT_DEFAULT (rtpmp4vpay_debug)
/* elementfactory information */
static const GstElementDetails gst_rtp_mp4vpay_details =
GST_ELEMENT_DETAILS ("RTP packet payloader",
"Codec/Payloader/Network",
"Payload MPEG4 video as RTP packets (RFC 3016)",
"Wim Taymans <wim@fluendo.com>");
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")
);
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_SEND_CONFIG FALSE
enum
{
ARG_0,
ARG_SEND_CONFIG
};
static void gst_rtp_mp4v_pay_class_init (GstRtpMP4VPayClass * klass);
static void gst_rtp_mp4v_pay_base_init (GstRtpMP4VPayClass * klass);
static void gst_rtp_mp4v_pay_init (GstRtpMP4VPay * rtpmp4vpay);
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 (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_mp4v_pay_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
static gboolean gst_rtp_mp4v_pay_event (GstPad * pad, GstEvent * event);
static GstBaseRTPPayloadClass *parent_class = NULL;
static GType
gst_rtp_mp4v_pay_get_type (void)
{
static GType rtpmp4vpay_type = 0;
if (!rtpmp4vpay_type) {
static const GTypeInfo rtpmp4vpay_info = {
sizeof (GstRtpMP4VPayClass),
(GBaseInitFunc) gst_rtp_mp4v_pay_base_init,
NULL,
(GClassInitFunc) gst_rtp_mp4v_pay_class_init,
NULL,
NULL,
sizeof (GstRtpMP4VPay),
0,
(GInstanceInitFunc) gst_rtp_mp4v_pay_init,
};
rtpmp4vpay_type =
g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpMP4VPay",
&rtpmp4vpay_info, 0);
}
return rtpmp4vpay_type;
}
static void
gst_rtp_mp4v_pay_base_init (GstRtpMP4VPayClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_mp4v_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_mp4v_pay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_mp4vpay_details);
}
static void
gst_rtp_mp4v_pay_class_init (GstRtpMP4VPayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->set_property = gst_rtp_mp4v_pay_set_property;
gobject_class->get_property = gst_rtp_mp4v_pay_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SEND_CONFIG,
g_param_spec_boolean ("send-config", "Send Config",
"Send the config parameters in RTP packets as well",
DEFAULT_SEND_CONFIG, G_PARAM_READWRITE));
gobject_class->finalize = gst_rtp_mp4v_pay_finalize;
gstbasertppayload_class->set_caps = gst_rtp_mp4v_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_mp4v_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtpmp4vpay_debug, "rtpmp4vpay", 0,
"MP4 video RTP Payloader");
}
static void
gst_rtp_mp4v_pay_init (GstRtpMP4VPay * rtpmp4vpay)
{
GstPad *sinkpad;
rtpmp4vpay->adapter = gst_adapter_new ();
rtpmp4vpay->rate = 90000;
rtpmp4vpay->profile = 1;
rtpmp4vpay->send_config = DEFAULT_SEND_CONFIG;
sinkpad = GST_BASE_RTP_PAYLOAD_SINKPAD (rtpmp4vpay);
rtpmp4vpay->old_event_func = sinkpad->eventfunc;
gst_pad_set_event_function (sinkpad, gst_rtp_mp4v_pay_event);
}
static void
gst_rtp_mp4v_pay_finalize (GObject * object)
{
GstRtpMP4VPay *rtpmp4vpay;
rtpmp4vpay = GST_RTP_MP4V_PAY (object);
g_object_unref (rtpmp4vpay->adapter);
rtpmp4vpay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_rtp_mp4v_pay_new_caps (GstRtpMP4VPay * rtpmp4vpay)
{
gchar *profile, *config;
GValue v = { 0 };
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);
gst_basertppayload_set_outcaps (GST_BASE_RTP_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);
}
static gboolean
gst_rtp_mp4v_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
GstRtpMP4VPay *rtpmp4vpay;
GstStructure *structure;
const GValue *codec_data;
rtpmp4vpay = GST_RTP_MP4V_PAY (payload);
gst_basertppayload_set_options (payload, "video", TRUE, "MP4V-ES",
rtpmp4vpay->rate);
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;
guint8 *data;
guint size;
buffer = gst_value_get_buffer (codec_data);
data = GST_BUFFER_DATA (buffer);
size = GST_BUFFER_SIZE (buffer);
if (size < 5)
goto done;
rtpmp4vpay->profile = data[4];
GST_LOG_OBJECT (rtpmp4vpay, "configuring codec_data, profile %d",
data[4]);
if (rtpmp4vpay->config)
gst_buffer_unref (rtpmp4vpay->config);
rtpmp4vpay->config = gst_buffer_copy (buffer);
gst_rtp_mp4v_pay_new_caps (rtpmp4vpay);
}
}
done:
return TRUE;
}
static void
gst_rtp_mp4v_pay_empty (GstRtpMP4VPay * rtpmp4vpay)
{
gst_adapter_clear (rtpmp4vpay->adapter);
}
static GstFlowReturn
gst_rtp_mp4v_pay_flush (GstRtpMP4VPay * rtpmp4vpay)
{
guint avail;
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 MP4V data
* over multiple packets. */
avail = gst_adapter_available (rtpmp4vpay->adapter);
ret = GST_FLOW_OK;
while (avail > 0) {
guint towrite;
guint8 *payload;
guint payload_len;
guint packet_len;
/* 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, GST_BASE_RTP_PAYLOAD_MTU (rtpmp4vpay));
/* 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);
/* copy payload */
payload = gst_rtp_buffer_get_payload (outbuf);
gst_adapter_copy (rtpmp4vpay->adapter, payload, 0, payload_len);
gst_adapter_flush (rtpmp4vpay->adapter, payload_len);
avail -= payload_len;
gst_rtp_buffer_set_marker (outbuf, avail == 0);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4vpay->first_ts;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4vpay), outbuf);
}
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)
{
guint32 code;
gboolean result;
*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:
{
gint i;
guint8 profile;
gboolean newprofile = FALSE;
gboolean equal;
/* 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_SIZE (enc->config) == i) {
equal = memcmp (GST_BUFFER_DATA (enc->config), 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);
memcpy (GST_BUFFER_DATA (enc->config), 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;
break;
default:
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 (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpMP4VPay *rtpmp4vpay;
GstFlowReturn ret;
guint size, avail;
guint packet_len;
guint8 *data;
gboolean flush;
gint strip;
GstClockTime duration;
ret = GST_FLOW_OK;
rtpmp4vpay = GST_RTP_MP4V_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
duration = GST_BUFFER_DURATION (buffer);
avail = gst_adapter_available (rtpmp4vpay->adapter);
/* empty buffer, take timestamp */
if (avail == 0) {
rtpmp4vpay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
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, data, size, &strip);
if (strip) {
/* strip off config if requested */
if (!rtpmp4vpay->send_config) {
GstBuffer *subbuf;
/* strip off header */
subbuf = gst_buffer_create_sub (buffer, strip, size - strip);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buffer);
gst_buffer_unref (buffer);
buffer = subbuf;
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
}
}
/* if we need to flush, do so now */
if (flush) {
ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay);
rtpmp4vpay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
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_basertppayload_is_filled (basepayload,
packet_len, rtpmp4vpay->duration + duration)) {
ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay);
rtpmp4vpay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtpmp4vpay->duration = 0;
}
/* push new data */
gst_adapter_push (rtpmp4vpay->adapter, buffer);
rtpmp4vpay->duration += duration;
return ret;
}
static gboolean
gst_rtp_mp4v_pay_event (GstPad * pad, GstEvent * event)
{
GstRtpMP4VPay *rtpmp4vpay;
gboolean ret;
rtpmp4vpay = GST_RTP_MP4V_PAY (gst_pad_get_parent (pad));
GST_DEBUG ("Got event: %s", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
gst_rtp_mp4v_pay_flush (rtpmp4vpay);
break;
case GST_EVENT_FLUSH_STOP:
gst_rtp_mp4v_pay_empty (rtpmp4vpay);
break;
default:
break;
}
ret = rtpmp4vpay->old_event_func (pad, event);
g_object_unref (rtpmp4vpay);
return ret;
}
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 ARG_SEND_CONFIG:
rtpmp4vpay->send_config = g_value_get_boolean (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 ARG_SEND_CONFIG:
g_value_set_boolean (value, rtpmp4vpay->send_config);
break;
default:
break;
}
}
gboolean
gst_rtp_mp4v_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpmp4vpay",
GST_RANK_NONE, GST_TYPE_RTP_MP4V_PAY);
}