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gstreamer/gst/rtp/gstrtph264pay.c
Wim Taymans 1f6b06ce66 rtph264pay: scale spspps_interval to milliseconds 
The spspps_interval is kept in seconds. Convert it to milliseconds before
comparing it to another value in milliseconds.
2010-01-20 16:29:57 +01:00

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/* ex: set tabstop=2 shiftwidth=2 expandtab: */
/* GStreamer
*
* 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 "gstrtph264pay.h"
#define IDR_TYPE_ID 5
#define SPS_TYPE_ID 7
#define PPS_TYPE_ID 8
#define USE_MEMCMP
GST_DEBUG_CATEGORY_STATIC (rtph264pay_debug);
#define GST_CAT_DEFAULT (rtph264pay_debug)
/* references:
*
* RFC 3984
*/
/* elementfactory information */
static const GstElementDetails gst_rtp_h264pay_details =
GST_ELEMENT_DETAILS ("RTP H264 payloader",
"Codec/Payloader/Network",
"Payload-encode H264 video into RTP packets (RFC 3984)",
"Laurent Glayal <spglegle@yahoo.fr>");
static GstStaticPadTemplate gst_rtp_h264_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/x-h264")
);
static GstStaticPadTemplate gst_rtp_h264_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) 90000, " "encoding-name = (string) \"H264\"")
);
#define GST_TYPE_H264_SCAN_MODE (gst_h264_scan_mode_get_type())
static GType
gst_h264_scan_mode_get_type (void)
{
static GType h264_scan_mode_type = 0;
static const GEnumValue h264_scan_modes[] = {
{GST_H264_SCAN_MODE_BYTESTREAM,
"Scan complete bytestream for NALUs (not implemented)",
"bytestream"},
{GST_H264_SCAN_MODE_MULTI_NAL, "Buffers contain multiple complete NALUs",
"multiple"},
{GST_H264_SCAN_MODE_SINGLE_NAL, "Buffers contain a single complete NALU",
"single"},
{0, NULL, NULL},
};
if (!h264_scan_mode_type) {
h264_scan_mode_type =
g_enum_register_static ("GstH264PayScanMode", h264_scan_modes);
}
return h264_scan_mode_type;
}
#define DEFAULT_PROFILE_LEVEL_ID NULL
#define DEFAULT_SPROP_PARAMETER_SETS NULL
#define DEFAULT_SCAN_MODE GST_H264_SCAN_MODE_MULTI_NAL
#define DEFAULT_BUFFER_LIST FALSE
#define DEFAULT_CONFIG_INTERVAL 0
enum
{
PROP_0,
PROP_PROFILE_LEVEL_ID,
PROP_SPROP_PARAMETER_SETS,
PROP_SCAN_MODE,
PROP_BUFFER_LIST,
PROP_CONFIG_INTERVAL,
PROP_LAST
};
#define IS_ACCESS_UNIT(x) (((x) > 0x00) && ((x) < 0x06))
static void gst_rtp_h264_pay_finalize (GObject * object);
static void gst_rtp_h264_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_h264_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload,
GstCaps * caps);
static GstFlowReturn gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * pad,
GstBuffer * buffer);
GST_BOILERPLATE (GstRtpH264Pay, gst_rtp_h264_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_rtp_h264_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_h264_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_h264_pay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_h264pay_details);
}
static void
gst_rtp_h264_pay_class_init (GstRtpH264PayClass * klass)
{
GObjectClass *gobject_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
gobject_class->set_property = gst_rtp_h264_pay_set_property;
gobject_class->get_property = gst_rtp_h264_pay_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_PROFILE_LEVEL_ID, g_param_spec_string ("profile-level-id",
"profile-level-id",
"The base64 profile-level-id to set in the sink caps (deprecated)",
DEFAULT_PROFILE_LEVEL_ID,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_SPROP_PARAMETER_SETS, g_param_spec_string ("sprop-parameter-sets",
"sprop-parameter-sets",
"The base64 sprop-parameter-sets to set in out caps (set to NULL to "
"extract from stream)",
DEFAULT_SPROP_PARAMETER_SETS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SCAN_MODE,
g_param_spec_enum ("scan-mode", "Scan Mode",
"How to scan the input buffers for NAL units. Performance can be "
"increased when certain assumptions are made about the input buffers",
GST_TYPE_H264_SCAN_MODE, DEFAULT_SCAN_MODE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_LIST,
g_param_spec_boolean ("buffer-list", "Buffer List",
"Use Buffer Lists",
DEFAULT_BUFFER_LIST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_CONFIG_INTERVAL,
g_param_spec_uint ("config-interval",
"SPS PPS Send Interval",
"Send SPS and PPS Insertion Interval in seconds (sprop parameter sets "
"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_h264_pay_finalize;
gstbasertppayload_class->set_caps = gst_rtp_h264_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_h264_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtph264pay_debug, "rtph264pay", 0,
"H264 RTP Payloader");
}
static void
gst_rtp_h264_pay_init (GstRtpH264Pay * rtph264pay, GstRtpH264PayClass * klass)
{
rtph264pay->queue = g_array_new (FALSE, FALSE, sizeof (guint));
rtph264pay->profile = 0;
rtph264pay->sps = NULL;
rtph264pay->pps = NULL;
rtph264pay->last_spspps = -1;
rtph264pay->scan_mode = GST_H264_SCAN_MODE_MULTI_NAL;
rtph264pay->buffer_list = DEFAULT_BUFFER_LIST;
rtph264pay->spspps_interval = DEFAULT_CONFIG_INTERVAL;
}
static void
gst_rtp_h264_pay_clear_sps_pps (GstRtpH264Pay * rtph264pay)
{
g_list_foreach (rtph264pay->sps, (GFunc) gst_mini_object_unref, NULL);
g_list_free (rtph264pay->sps);
rtph264pay->sps = NULL;
g_list_foreach (rtph264pay->pps, (GFunc) gst_mini_object_unref, NULL);
g_list_free (rtph264pay->pps);
rtph264pay->pps = NULL;
}
static void
gst_rtp_h264_pay_finalize (GObject * object)
{
GstRtpH264Pay *rtph264pay;
rtph264pay = GST_RTP_H264_PAY (object);
g_array_free (rtph264pay->queue, TRUE);
gst_rtp_h264_pay_clear_sps_pps (rtph264pay);
g_free (rtph264pay->sprop_parameter_sets);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
/* take the currently configured SPS and PPS lists and set them on the caps as
* sprop-parameter-sets */
static gboolean
gst_rtp_h264_pay_set_sps_pps (GstBaseRTPPayload * basepayload)
{
GstRtpH264Pay *payloader = GST_RTP_H264_PAY (basepayload);
gchar *profile;
gchar *set;
GList *walk;
GString *sprops;
guint count;
gboolean res;
sprops = g_string_new ("");
count = 0;
/* build the sprop-parameter-sets */
for (walk = payloader->sps; walk; walk = g_list_next (walk)) {
GstBuffer *sps_buf = GST_BUFFER_CAST (walk->data);
set =
g_base64_encode (GST_BUFFER_DATA (sps_buf), GST_BUFFER_SIZE (sps_buf));
g_string_append_printf (sprops, "%s%s", count ? "," : "", set);
g_free (set);
count++;
}
for (walk = payloader->pps; walk; walk = g_list_next (walk)) {
GstBuffer *pps_buf = GST_BUFFER_CAST (walk->data);
set =
g_base64_encode (GST_BUFFER_DATA (pps_buf), GST_BUFFER_SIZE (pps_buf));
g_string_append_printf (sprops, "%s%s", count ? "," : "", set);
g_free (set);
count++;
}
/* profile is 24 bit. Force it to respect the limit */
profile = g_strdup_printf ("%06x", payloader->profile & 0xffffff);
/* combine into output caps */
res = gst_basertppayload_set_outcaps (basepayload,
"sprop-parameter-sets", G_TYPE_STRING, sprops->str, NULL);
g_string_free (sprops, TRUE);
g_free (profile);
return res;
}
static gboolean
gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
{
GstRtpH264Pay *rtph264pay;
GstStructure *str;
const GValue *value;
guint8 *data;
guint size;
rtph264pay = GST_RTP_H264_PAY (basepayload);
str = gst_caps_get_structure (caps, 0);
/* we can only set the output caps when we found the sprops and profile
* NALs */
gst_basertppayload_set_options (basepayload, "video", TRUE, "H264", 90000);
/* packetized AVC video has a codec_data */
if ((value = gst_structure_get_value (str, "codec_data"))) {
GstBuffer *buffer;
guint num_sps, num_pps;
gint i, nal_size;
GST_DEBUG_OBJECT (rtph264pay, "have packetized h264");
rtph264pay->packetized = TRUE;
buffer = gst_value_get_buffer (value);
data = GST_BUFFER_DATA (buffer);
size = GST_BUFFER_SIZE (buffer);
/* parse the avcC data */
if (size < 7)
goto avcc_too_small;
/* parse the version, this must be 1 */
if (data[0] != 1)
goto wrong_version;
/* AVCProfileIndication */
/* profile_compat */
/* AVCLevelIndication */
rtph264pay->profile = (data[1] << 16) | (data[2] << 8) | data[3];
GST_DEBUG_OBJECT (rtph264pay, "profile %06x", rtph264pay->profile);
/* 6 bits reserved | 2 bits lengthSizeMinusOne */
/* this is the number of bytes in front of the NAL units to mark their
* length */
rtph264pay->nal_length_size = (data[4] & 0x03) + 1;
GST_DEBUG_OBJECT (rtph264pay, "nal length %u", rtph264pay->nal_length_size);
/* 3 bits reserved | 5 bits numOfSequenceParameterSets */
num_sps = data[5] & 0x1f;
GST_DEBUG_OBJECT (rtph264pay, "num SPS %u", num_sps);
data += 6;
size -= 6;
/* create the sprop-parameter-sets */
for (i = 0; i < num_sps; i++) {
GstBuffer *sps_buf;
if (size < 2)
goto avcc_error;
nal_size = (data[0] << 8) | data[1];
data += 2;
size -= 2;
GST_LOG_OBJECT (rtph264pay, "SPS %d size %d", i, nal_size);
if (size < nal_size)
goto avcc_error;
/* make a buffer out of it and add to SPS list */
sps_buf = gst_buffer_new_and_alloc (nal_size);
memcpy (GST_BUFFER_DATA (sps_buf), data, nal_size);
rtph264pay->sps = g_list_append (rtph264pay->sps, sps_buf);
data += nal_size;
size -= nal_size;
}
if (size < 1)
goto avcc_error;
/* 8 bits numOfPictureParameterSets */
num_pps = data[0];
data += 1;
size -= 1;
GST_DEBUG_OBJECT (rtph264pay, "num PPS %u", num_pps);
for (i = 0; i < num_pps; i++) {
GstBuffer *pps_buf;
if (size < 2)
goto avcc_error;
nal_size = (data[0] << 8) | data[1];
data += 2;
size -= 2;
GST_LOG_OBJECT (rtph264pay, "PPS %d size %d", i, nal_size);
if (size < nal_size)
goto avcc_error;
/* make a buffer out of it and add to PPS list */
pps_buf = gst_buffer_new_and_alloc (nal_size);
memcpy (GST_BUFFER_DATA (pps_buf), data, nal_size);
rtph264pay->pps = g_list_append (rtph264pay->pps, pps_buf);
data += nal_size;
size -= nal_size;
}
/* and update the caps with the collected data */
if (!gst_rtp_h264_pay_set_sps_pps (basepayload))
return FALSE;
} else {
GST_DEBUG_OBJECT (rtph264pay, "have bytestream h264");
rtph264pay->packetized = FALSE;
}
return TRUE;
avcc_too_small:
{
GST_ERROR_OBJECT (rtph264pay, "avcC size %u < 7", size);
return FALSE;
}
wrong_version:
{
GST_ERROR_OBJECT (rtph264pay, "wrong avcC version");
return FALSE;
}
avcc_error:
{
GST_ERROR_OBJECT (rtph264pay, "avcC too small ");
return FALSE;
}
}
static guint
next_start_code (guint8 * data, guint size)
{
/* Boyer-Moore string matching algorithm, in a degenerative
* sense because our search 'alphabet' is binary - 0 & 1 only.
* This allow us to simplify the general BM algorithm to a very
* simple form. */
/* assume 1 is in the 4th byte */
guint offset = 3;
while (offset < size) {
if (1 == data[offset]) {
unsigned int shift = offset;
if (0 == data[--shift]) {
if (0 == data[--shift]) {
if (0 == data[--shift]) {
return shift;
}
}
}
/* The jump is always 4 because of the 1 previously matched.
* All the 0's must be after this '1' matched at offset */
offset += 4;
} else if (0 == data[offset]) {
/* maybe next byte is 1? */
offset++;
} else {
/* can jump 4 bytes forward */
offset += 4;
}
/* at each iteration, we rescan in a backward manner until
* we match 0.0.0.1 in reverse order. Since our search string
* has only 2 'alpabets' (i.e. 0 & 1), we know that any
* mismatch will force us to shift a fixed number of steps */
}
GST_DEBUG ("Cannot find next NAL start code. returning %u", size);
return size;
}
static gboolean
gst_rtp_h264_pay_decode_nal (GstRtpH264Pay * payloader,
guint8 * data, guint size, GstClockTime timestamp)
{
guint8 *sps = NULL, *pps = NULL;
guint sps_len = 0, pps_len = 0;
guint8 header, type;
guint len;
gboolean updated;
/* default is no update */
updated = FALSE;
GST_DEBUG ("NAL payload len=%u", size);
len = size;
header = data[0];
type = header & 0x1f;
/* keep sps & pps separately so that we can update either one
* independently. We also record the timestamp of the last SPS/PPS so
* that we can insert them at regular intervals and when needed. */
if (SPS_TYPE_ID == type) {
/* encode the entire SPS NAL in base64 */
GST_DEBUG ("Found SPS %x %x %x Len=%u", (header >> 7),
(header >> 5) & 3, type, len);
sps = data;
sps_len = len;
/* remember when we last saw SPS */
if (timestamp != -1)
payloader->last_spspps = timestamp;
} else if (PPS_TYPE_ID == type) {
/* encoder the entire PPS NAL in base64 */
GST_DEBUG ("Found PPS %x %x %x Len = %u",
(header >> 7), (header >> 5) & 3, type, len);
pps = data;
pps_len = len;
/* remember when we last saw PPS */
if (timestamp != -1)
payloader->last_spspps = timestamp;
} else {
GST_DEBUG ("NAL: %x %x %x Len = %u", (header >> 7),
(header >> 5) & 3, type, len);
}
/* If we encountered an SPS and/or a PPS, check if it's the
* same as the one we have. If not, update our version and
* set updated to TRUE
*/
if (sps_len > 0) {
GstBuffer *sps_buf;
if (payloader->sps != NULL) {
sps_buf = GST_BUFFER_CAST (payloader->sps->data);
if ((GST_BUFFER_SIZE (sps_buf) != sps_len)
|| memcmp (GST_BUFFER_DATA (sps_buf), sps, sps_len)) {
/* something changed, update */
payloader->profile = (sps[1] << 16) + (sps[2] << 8) + sps[3];
GST_DEBUG ("Profile level IDC = %06x", payloader->profile);
updated = TRUE;
}
} else {
/* no previous SPS, update */
updated = TRUE;
}
if (updated) {
sps_buf = gst_buffer_new_and_alloc (sps_len);
memcpy (GST_BUFFER_DATA (sps_buf), sps, sps_len);
if (payloader->sps) {
/* replace old buffer */
gst_buffer_unref (payloader->sps->data);
payloader->sps->data = sps_buf;
} else {
/* add new buffer */
payloader->sps = g_list_prepend (payloader->sps, sps_buf);
}
}
}
if (pps_len > 0) {
GstBuffer *pps_buf;
if (payloader->pps != NULL) {
pps_buf = GST_BUFFER_CAST (payloader->pps->data);
if ((GST_BUFFER_SIZE (pps_buf) != pps_len)
|| memcmp (GST_BUFFER_DATA (pps_buf), pps, pps_len)) {
/* something changed, update */
updated = TRUE;
}
} else {
/* no previous SPS, update */
updated = TRUE;
}
if (updated) {
pps_buf = gst_buffer_new_and_alloc (pps_len);
memcpy (GST_BUFFER_DATA (pps_buf), pps, pps_len);
if (payloader->pps) {
/* replace old buffer */
gst_buffer_unref (payloader->pps->data);
payloader->pps->data = pps_buf;
} else {
/* add new buffer */
payloader->pps = g_list_prepend (payloader->pps, pps_buf);
}
}
}
return updated;
}
static GstFlowReturn
gst_rtp_h264_pay_payload_nal (GstBaseRTPPayload * basepayload, guint8 * data,
guint size, GstClockTime timestamp, GstBuffer * buffer_orig);
static GstFlowReturn
gst_rtp_h264_pay_send_sps_pps (GstBaseRTPPayload * basepayload,
GstRtpH264Pay * rtph264pay, GstClockTime timestamp)
{
GstFlowReturn ret = GST_FLOW_OK;
GList *walk;
for (walk = rtph264pay->sps; walk; walk = g_list_next (walk)) {
GstBuffer *sps_buf = GST_BUFFER_CAST (walk->data);
GST_DEBUG_OBJECT (rtph264pay, "inserting SPS in the stream");
/* resend SPS */
ret = gst_rtp_h264_pay_payload_nal (basepayload,
GST_BUFFER_DATA (sps_buf), GST_BUFFER_SIZE (sps_buf), timestamp,
sps_buf);
/* Not critical here; but throw a warning */
if (ret != GST_FLOW_OK)
GST_WARNING ("Problem pushing SPS");
}
for (walk = rtph264pay->pps; walk; walk = g_list_next (walk)) {
GstBuffer *pps_buf = GST_BUFFER_CAST (walk->data);
GST_DEBUG_OBJECT (rtph264pay, "inserting PPS in the stream");
/* resend PPS */
ret = gst_rtp_h264_pay_payload_nal (basepayload,
GST_BUFFER_DATA (pps_buf), GST_BUFFER_SIZE (pps_buf), timestamp,
pps_buf);
/* Not critical here; but throw a warning */
if (ret != GST_FLOW_OK)
GST_WARNING ("Problem pushing PPS");
}
if (timestamp != -1)
rtph264pay->last_spspps = timestamp;
return ret;
}
static GstFlowReturn
gst_rtp_h264_pay_payload_nal (GstBaseRTPPayload * basepayload, guint8 * data,
guint size, GstClockTime timestamp, GstBuffer * buffer_orig)
{
GstRtpH264Pay *rtph264pay;
GstFlowReturn ret;
guint8 nalType;
guint packet_len, payload_len, mtu;
GstBuffer *outbuf;
guint8 *payload;
GstBufferList *list = NULL;
GstBufferListIterator *it = NULL;
gboolean send_spspps;
rtph264pay = GST_RTP_H264_PAY (basepayload);
mtu = GST_BASE_RTP_PAYLOAD_MTU (rtph264pay);
nalType = data[0] & 0x1f;
GST_DEBUG_OBJECT (rtph264pay, "Processing Buffer with NAL TYPE=%d", nalType);
send_spspps = FALSE;
/* check if we need to emit an SPS/PPS now */
if (nalType == IDR_TYPE_ID && rtph264pay->spspps_interval > 0) {
if (rtph264pay->last_spspps != -1) {
guint diff;
/* calculate diff between last SPS/PPS in milliseconds */
if (timestamp > rtph264pay->last_spspps)
diff = GST_TIME_AS_MSECONDS (timestamp - rtph264pay->last_spspps);
else
diff = 0;
GST_DEBUG_OBJECT (rtph264pay, "interval since last SPS/PPS %ums", diff);
/* bigger than interval, queue SPS/PPS */
if (diff >= (rtph264pay->spspps_interval * GST_MSECOND)) {
GST_DEBUG_OBJECT (rtph264pay, "time to send SPS/PPS");
send_spspps = TRUE;
}
} else {
/* no know previous SPS/PPS time, send now */
GST_DEBUG_OBJECT (rtph264pay, "no previous SPS/PPS time, send now");
send_spspps = TRUE;
}
}
if (send_spspps) {
/* we need to send SPS/PPS now first. FIXME, don't use the timestamp for
* checking when we need to send SPS/PPS but convert to running_time first. */
ret = gst_rtp_h264_pay_send_sps_pps (basepayload, rtph264pay, timestamp);
if (ret != GST_FLOW_OK)
return ret;
}
packet_len = gst_rtp_buffer_calc_packet_len (size, 0, 0);
if (packet_len < mtu) {
GST_DEBUG_OBJECT (basepayload,
"NAL Unit fit in one packet datasize=%d mtu=%d", size, mtu);
/* will fit in one packet */
if (rtph264pay->buffer_list) {
/* use buffer lists
* first create buffer without payload containing only the RTP header
* and then another buffer containing the payload. both buffers will
* be then added to the list */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
} else {
/* use the old-fashioned way with a single buffer and memcpy */
outbuf = gst_rtp_buffer_new_allocate (size, 0, 0);
}
/* only set the marker bit on packets containing access units */
if (IS_ACCESS_UNIT (nalType)) {
gst_rtp_buffer_set_marker (outbuf, 1);
}
/* timestamp the outbuffer */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
if (rtph264pay->buffer_list) {
GstBuffer *paybuf;
/* create another buffer with the payload. */
if (buffer_orig)
paybuf = gst_buffer_create_sub (buffer_orig, data -
GST_BUFFER_DATA (buffer_orig), size);
else {
paybuf = gst_buffer_new_and_alloc (size);
memcpy (GST_BUFFER_DATA (paybuf), data, size);
}
list = gst_buffer_list_new ();
it = gst_buffer_list_iterate (list);
/* add both buffers to the buffer list */
gst_buffer_list_iterator_add_group (it);
gst_buffer_list_iterator_add (it, outbuf);
gst_buffer_list_iterator_add (it, paybuf);
gst_buffer_list_iterator_free (it);
/* push the list to the next element in the pipe */
ret = gst_basertppayload_push_list (basepayload, list);
} else {
payload = gst_rtp_buffer_get_payload (outbuf);
GST_DEBUG_OBJECT (basepayload, "Copying %d bytes to outbuf", size);
memcpy (payload, data, size);
ret = gst_basertppayload_push (basepayload, outbuf);
}
} else {
/* fragmentation Units FU-A */
guint8 nalHeader;
guint limitedSize;
int ii = 0, start = 1, end = 0, pos = 0;
GST_DEBUG_OBJECT (basepayload,
"NAL Unit DOES NOT fit in one packet datasize=%d mtu=%d", size, mtu);
nalHeader = *data;
pos++;
size--;
ret = GST_FLOW_OK;
GST_DEBUG_OBJECT (basepayload, "Using FU-A fragmentation for data size=%d",
size);
/* We keep 2 bytes for FU indicator and FU Header */
payload_len = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0);
if (rtph264pay->buffer_list) {
list = gst_buffer_list_new ();
it = gst_buffer_list_iterate (list);
}
while (end == 0) {
limitedSize = size < payload_len ? size : payload_len;
GST_DEBUG_OBJECT (basepayload,
"Inside FU-A fragmentation limitedSize=%d iteration=%d", limitedSize,
ii);
if (rtph264pay->buffer_list) {
/* use buffer lists
* first create buffer without payload containing only the RTP header
* and then another buffer containing the payload. both buffers will
* be then added to the list */
outbuf = gst_rtp_buffer_new_allocate (2, 0, 0);
} else {
/* use the old-fashioned way with a single buffer and memcpy
* first create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (limitedSize + 2, 0, 0);
}
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
payload = gst_rtp_buffer_get_payload (outbuf);
if (limitedSize == size) {
GST_DEBUG_OBJECT (basepayload, "end size=%d iteration=%d", size, ii);
end = 1;
}
if (IS_ACCESS_UNIT (nalType)) {
gst_rtp_buffer_set_marker (outbuf, end);
}
/* FU indicator */
payload[0] = (nalHeader & 0x60) | 28;
/* FU Header */
payload[1] = (start << 7) | (end << 6) | (nalHeader & 0x1f);
if (rtph264pay->buffer_list) {
GstBuffer *paybuf;
/* create another buffer to hold the payload */
if (buffer_orig)
paybuf = gst_buffer_create_sub (buffer_orig, data -
GST_BUFFER_DATA (buffer_orig) + pos, limitedSize);
else {
paybuf = gst_buffer_new_and_alloc (limitedSize);
memcpy (GST_BUFFER_DATA (paybuf), data + pos, limitedSize);
}
/* create a new group to hold the header and the payload */
gst_buffer_list_iterator_add_group (it);
/* add both buffers to the buffer list */
gst_buffer_list_iterator_add (it, outbuf);
gst_buffer_list_iterator_add (it, paybuf);
} else {
memcpy (&payload[2], data + pos, limitedSize);
GST_DEBUG_OBJECT (basepayload,
"recorded %d payload bytes into packet iteration=%d",
limitedSize + 2, ii);
ret = gst_basertppayload_push (basepayload, outbuf);
if (ret != GST_FLOW_OK)
break;
}
size -= limitedSize;
pos += limitedSize;
ii++;
start = 0;
}
if (rtph264pay->buffer_list) {
/* free iterator and push the whole buffer list at once */
gst_buffer_list_iterator_free (it);
ret = gst_basertppayload_push_list (basepayload, list);
}
}
return ret;
}
static GstFlowReturn
gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpH264Pay *rtph264pay;
GstFlowReturn ret;
guint size, nal_len, i;
guint8 *data, *nal_data;
GstClockTime timestamp;
GArray *nal_queue;
rtph264pay = GST_RTP_H264_PAY (basepayload);
/* the input buffer contains one or more NAL units */
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
ret = GST_FLOW_OK;
GST_DEBUG_OBJECT (basepayload, "got %d bytes", size);
/* now loop over all NAL units and put them in a packet
* FIXME, we should really try to pack multiple NAL units into one RTP packet
* if we can, especially for the config packets that wont't cause decoder
* latency. */
if (rtph264pay->packetized) {
guint nal_length_size;
nal_length_size = rtph264pay->nal_length_size;
while (size > nal_length_size) {
gint i;
nal_len = 0;
for (i = 0; i < nal_length_size; i++) {
nal_len = ((nal_len << 8) + data[i]);
}
/* skip the length bytes, make sure we don't run past the buffer size */
data += nal_length_size;
size -= nal_length_size;
if (size >= nal_len) {
GST_DEBUG_OBJECT (basepayload, "got NAL of size %u", nal_len);
} else {
nal_len = size;
GST_DEBUG_OBJECT (basepayload, "got incomplete NAL of size %u",
nal_len);
}
ret =
gst_rtp_h264_pay_payload_nal (basepayload, data, nal_len, timestamp,
buffer);
if (ret != GST_FLOW_OK)
break;
data += nal_len;
size -= nal_len;
}
} else {
guint next;
gboolean update = FALSE;
/* get offset of first start code */
next = next_start_code (data, size);
/* skip to start code, if no start code is found, next will be size and we
* will not collect data. */
data += next;
size -= next;
nal_data = data;
nal_queue = rtph264pay->queue;
/* array must be empty when we get here */
g_assert (nal_queue->len == 0);
GST_DEBUG_OBJECT (basepayload, "found first start at %u, bytes left %u",
next, size);
/* first pass to locate NALs and parse SPS/PPS */
while (size > 4) {
/* skip start code */
data += 4;
size -= 4;
if (rtph264pay->scan_mode == GST_H264_SCAN_MODE_SINGLE_NAL) {
/* we are told that there is only a single NAL in this packet so that we
* can avoid scanning for the next NAL. */
next = size;
} else {
/* use next_start_code() to scan buffer.
* next_start_code() returns the offset in data,
* starting from zero to the first byte of 0.0.0.1
* If no start code is found, it returns the value of the
* 'size' parameter.
* data is unchanged by the call to next_start_code()
*/
next = next_start_code (data, size);
}
/* nal length is distance to next start code */
nal_len = next;
GST_DEBUG_OBJECT (basepayload, "found next start at %u of size %u", next,
nal_len);
if (rtph264pay->sprop_parameter_sets != NULL) {
/* explicitly set profile and sprop, use those */
if (rtph264pay->update_caps) {
if (!gst_basertppayload_set_outcaps (basepayload,
"sprop-parameter-sets", G_TYPE_STRING,
rtph264pay->sprop_parameter_sets, NULL))
goto caps_rejected;
rtph264pay->update_caps = FALSE;
GST_DEBUG ("outcaps udpate: sprop-parameter-sets=%s",
rtph264pay->sprop_parameter_sets);
}
} else {
/* We know our stream is a valid H264 NAL packet,
* go parse it for SPS/PPS to enrich the caps */
/* order: make sure to check nal */
update =
gst_rtp_h264_pay_decode_nal (rtph264pay, data, nal_len, timestamp)
|| update;
}
/* move to next NAL packet */
data += nal_len;
size -= nal_len;
g_array_append_val (nal_queue, nal_len);
}
/* if has new SPS & PPS, update the output caps */
if (G_UNLIKELY (update))
if (!gst_rtp_h264_pay_set_sps_pps (basepayload))
goto caps_rejected;
/* second pass to payload and push */
data = nal_data;
for (i = 0; i < nal_queue->len; i++) {
nal_len = g_array_index (nal_queue, guint, i);
/* skip start code */
data += 4;
/* put the data in one or more RTP packets */
ret =
gst_rtp_h264_pay_payload_nal (basepayload, data, nal_len, timestamp,
buffer);
if (ret != GST_FLOW_OK) {
break;
}
/* move to next NAL packet */
data += nal_len;
size -= nal_len;
}
g_array_set_size (nal_queue, 0);
}
gst_buffer_unref (buffer);
return ret;
caps_rejected:
GST_WARNING_OBJECT (basepayload, "Could not set outcaps");
g_array_set_size (nal_queue, 0);
gst_buffer_unref (buffer);
return GST_FLOW_NOT_NEGOTIATED;
}
static void
gst_rtp_h264_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpH264Pay *rtph264pay;
rtph264pay = GST_RTP_H264_PAY (object);
switch (prop_id) {
case PROP_PROFILE_LEVEL_ID:
break;
case PROP_SPROP_PARAMETER_SETS:
g_free (rtph264pay->sprop_parameter_sets);
rtph264pay->sprop_parameter_sets = g_value_dup_string (value);
rtph264pay->update_caps = TRUE;
break;
case PROP_SCAN_MODE:
rtph264pay->scan_mode = g_value_get_enum (value);
break;
case PROP_BUFFER_LIST:
rtph264pay->buffer_list = g_value_get_boolean (value);
break;
case PROP_CONFIG_INTERVAL:
rtph264pay->spspps_interval = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_h264_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpH264Pay *rtph264pay;
rtph264pay = GST_RTP_H264_PAY (object);
switch (prop_id) {
case PROP_PROFILE_LEVEL_ID:
break;
case PROP_SPROP_PARAMETER_SETS:
g_value_set_string (value, rtph264pay->sprop_parameter_sets);
break;
case PROP_SCAN_MODE:
g_value_set_enum (value, rtph264pay->scan_mode);
break;
case PROP_BUFFER_LIST:
g_value_set_boolean (value, rtph264pay->buffer_list);
break;
case PROP_CONFIG_INTERVAL:
g_value_set_uint (value, rtph264pay->spspps_interval);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
gboolean
gst_rtp_h264_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtph264pay",
GST_RANK_NONE, GST_TYPE_RTP_H264_PAY);
}
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