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gstreamer/gst/rtp/gstrtph264pay.c
Sebastian Dröge e9a0307b94 rtph26[45]pay: Change default aggregate-mode to "none" for backwards compatibility 
We didn't aggregate at all in previous versions and there are apparently
various RTP implementations that don't handle aggregation well at all.

As part of this also document that for RTSP it is recommended to keep it
set to "none" while for WebRTC it should be set to "zero-latency".

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/749

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/692>
2020-08-08 10:08:31 +03:00

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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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/pbutils/pbutils.h>
#include <gst/video/video.h>
/* Included to not duplicate gst_rtp_h264_add_sps_pps () */
#include "gstrtph264depay.h"
#include "gstrtph264pay.h"
#include "gstrtputils.h"
#include "gstbuffermemory.h"
#define IDR_TYPE_ID 5
#define SPS_TYPE_ID 7
#define PPS_TYPE_ID 8
#define AUD_TYPE_ID 9
#define STAP_A_TYPE_ID 24
#define FU_A_TYPE_ID 28
GST_DEBUG_CATEGORY_STATIC (rtph264pay_debug);
#define GST_CAT_DEFAULT (rtph264pay_debug)
#define GST_TYPE_RTP_H264_AGGREGATE_MODE \
(gst_rtp_h264_aggregate_mode_get_type ())
static GType
gst_rtp_h264_aggregate_mode_get_type (void)
{
static GType type = 0;
static const GEnumValue values[] = {
{GST_RTP_H264_AGGREGATE_NONE, "Do not aggregate NAL units", "none"},
{GST_RTP_H264_AGGREGATE_ZERO_LATENCY,
"Aggregate NAL units until a VCL unit is included", "zero-latency"},
{GST_RTP_H264_AGGREGATE_MAX_STAP,
"Aggregate all NAL units with the same timestamp (adds one frame of"
" latency)", "max-stap"},
{0, NULL, NULL},
};
if (!type) {
type = g_enum_register_static ("GstRtpH264AggregateMode", values);
}
return type;
}
/* references:
*
* RFC 3984
*/
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, "
"stream-format = (string) avc, alignment = (string) au;"
"video/x-h264, "
"stream-format = (string) byte-stream, alignment = (string) { nal, au }")
);
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 DEFAULT_SPROP_PARAMETER_SETS NULL
#define DEFAULT_CONFIG_INTERVAL 0
#define DEFAULT_AGGREGATE_MODE GST_RTP_H264_AGGREGATE_NONE
enum
{
PROP_0,
PROP_SPROP_PARAMETER_SETS,
PROP_CONFIG_INTERVAL,
PROP_AGGREGATE_MODE,
};
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 GstCaps *gst_rtp_h264_pay_getcaps (GstRTPBasePayload * payload,
GstPad * pad, GstCaps * filter);
static gboolean gst_rtp_h264_pay_setcaps (GstRTPBasePayload * basepayload,
GstCaps * caps);
static GstFlowReturn gst_rtp_h264_pay_handle_buffer (GstRTPBasePayload * pad,
GstBuffer * buffer);
static gboolean gst_rtp_h264_pay_sink_event (GstRTPBasePayload * payload,
GstEvent * event);
static GstStateChangeReturn gst_rtp_h264_pay_change_state (GstElement *
element, GstStateChange transition);
static gboolean gst_rtp_h264_pay_src_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static void gst_rtp_h264_pay_reset_bundle (GstRtpH264Pay * rtph264pay);
#define gst_rtp_h264_pay_parent_class parent_class
G_DEFINE_TYPE (GstRtpH264Pay, gst_rtp_h264_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static void
gst_rtp_h264_pay_class_init (GstRtpH264PayClass * 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_h264_pay_set_property;
gobject_class->get_property = gst_rtp_h264_pay_get_property;
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_PARAM_DEPRECATED));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_CONFIG_INTERVAL,
g_param_spec_int ("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, -1 = send with every IDR frame)",
-1, 3600, DEFAULT_CONFIG_INTERVAL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
);
/**
* GstRtpH264Pay:aggregate-mode
*
* Bundle suitable SPS/PPS NAL units into STAP-A aggregate packets.
*
* This can potentially reduce RTP packetization overhead but not all
* RTP implementations handle it correctly.
*
* For best compatibility, it is recommended to set this to "none" (the
* default) for RTSP and for WebRTC to "zero-latency".
*
* Since: 1.18
*/
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_AGGREGATE_MODE,
g_param_spec_enum ("aggregate-mode",
"Attempt to use aggregate packets",
"Bundle suitable SPS/PPS NAL units into STAP-A "
"aggregate packets",
GST_TYPE_RTP_H264_AGGREGATE_MODE,
DEFAULT_AGGREGATE_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
);
gobject_class->finalize = gst_rtp_h264_pay_finalize;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_h264_pay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_h264_pay_sink_template);
gst_element_class_set_static_metadata (gstelement_class, "RTP H264 payloader",
"Codec/Payloader/Network/RTP",
"Payload-encode H264 video into RTP packets (RFC 3984)",
"Laurent Glayal <spglegle@yahoo.fr>");
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_rtp_h264_pay_change_state);
gstrtpbasepayload_class->get_caps = gst_rtp_h264_pay_getcaps;
gstrtpbasepayload_class->set_caps = gst_rtp_h264_pay_setcaps;
gstrtpbasepayload_class->handle_buffer = gst_rtp_h264_pay_handle_buffer;
gstrtpbasepayload_class->sink_event = gst_rtp_h264_pay_sink_event;
GST_DEBUG_CATEGORY_INIT (rtph264pay_debug, "rtph264pay", 0,
"H264 RTP Payloader");
gst_type_mark_as_plugin_api (GST_TYPE_RTP_H264_AGGREGATE_MODE, 0);
}
static void
gst_rtp_h264_pay_init (GstRtpH264Pay * rtph264pay)
{
rtph264pay->queue = g_array_new (FALSE, FALSE, sizeof (guint));
rtph264pay->profile = 0;
rtph264pay->sps = g_ptr_array_new_with_free_func (
(GDestroyNotify) gst_buffer_unref);
rtph264pay->pps = g_ptr_array_new_with_free_func (
(GDestroyNotify) gst_buffer_unref);
rtph264pay->last_spspps = -1;
rtph264pay->spspps_interval = DEFAULT_CONFIG_INTERVAL;
rtph264pay->aggregate_mode = DEFAULT_AGGREGATE_MODE;
rtph264pay->delta_unit = FALSE;
rtph264pay->discont = FALSE;
rtph264pay->adapter = gst_adapter_new ();
gst_pad_set_query_function (GST_RTP_BASE_PAYLOAD_SRCPAD (rtph264pay),
gst_rtp_h264_pay_src_query);
}
static void
gst_rtp_h264_pay_clear_sps_pps (GstRtpH264Pay * rtph264pay)
{
g_ptr_array_set_size (rtph264pay->sps, 0);
g_ptr_array_set_size (rtph264pay->pps, 0);
}
static void
gst_rtp_h264_pay_finalize (GObject * object)
{
GstRtpH264Pay *rtph264pay;
rtph264pay = GST_RTP_H264_PAY (object);
g_array_free (rtph264pay->queue, TRUE);
g_ptr_array_free (rtph264pay->sps, TRUE);
g_ptr_array_free (rtph264pay->pps, TRUE);
g_free (rtph264pay->sprop_parameter_sets);
g_object_unref (rtph264pay->adapter);
gst_rtp_h264_pay_reset_bundle (rtph264pay);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static const gchar all_levels[][4] = {
"1",
"1b",
"1.1",
"1.2",
"1.3",
"2",
"2.1",
"2.2",
"3",
"3.1",
"3.2",
"4",
"4.1",
"4.2",
"5",
"5.1"
};
static GstCaps *
gst_rtp_h264_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad,
GstCaps * filter)
{
GstCaps *template_caps;
GstCaps *allowed_caps;
GstCaps *caps, *icaps;
gboolean append_unrestricted;
guint i;
allowed_caps =
gst_pad_peer_query_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload), NULL);
if (allowed_caps == NULL)
return NULL;
template_caps =
gst_static_pad_template_get_caps (&gst_rtp_h264_pay_sink_template);
if (gst_caps_is_any (allowed_caps)) {
caps = gst_caps_ref (template_caps);
goto done;
}
if (gst_caps_is_empty (allowed_caps)) {
caps = gst_caps_ref (allowed_caps);
goto done;
}
caps = gst_caps_new_empty ();
append_unrestricted = FALSE;
for (i = 0; i < gst_caps_get_size (allowed_caps); i++) {
GstStructure *s = gst_caps_get_structure (allowed_caps, i);
GstStructure *new_s = gst_structure_new_empty ("video/x-h264");
const gchar *profile_level_id;
profile_level_id = gst_structure_get_string (s, "profile-level-id");
if (profile_level_id && strlen (profile_level_id) == 6) {
const gchar *profile;
const gchar *level;
long int spsint;
guint8 sps[3];
spsint = strtol (profile_level_id, NULL, 16);
sps[0] = spsint >> 16;
sps[1] = spsint >> 8;
sps[2] = spsint;
profile = gst_codec_utils_h264_get_profile (sps, 3);
level = gst_codec_utils_h264_get_level (sps, 3);
if (profile && level) {
GST_LOG_OBJECT (payload, "In caps, have profile %s and level %s",
profile, level);
if (!strcmp (profile, "constrained-baseline"))
gst_structure_set (new_s, "profile", G_TYPE_STRING, profile, NULL);
else {
GValue val = { 0, };
GValue profiles = { 0, };
g_value_init (&profiles, GST_TYPE_LIST);
g_value_init (&val, G_TYPE_STRING);
g_value_set_static_string (&val, profile);
gst_value_list_append_value (&profiles, &val);
g_value_set_static_string (&val, "constrained-baseline");
gst_value_list_append_value (&profiles, &val);
gst_structure_take_value (new_s, "profile", &profiles);
}
if (!strcmp (level, "1"))
gst_structure_set (new_s, "level", G_TYPE_STRING, level, NULL);
else {
GValue levels = { 0, };
GValue val = { 0, };
int j;
g_value_init (&levels, GST_TYPE_LIST);
g_value_init (&val, G_TYPE_STRING);
for (j = 0; j < G_N_ELEMENTS (all_levels); j++) {
g_value_set_static_string (&val, all_levels[j]);
gst_value_list_prepend_value (&levels, &val);
if (!strcmp (level, all_levels[j]))
break;
}
gst_structure_take_value (new_s, "level", &levels);
}
} else {
/* Invalid profile-level-id means baseline */
gst_structure_set (new_s,
"profile", G_TYPE_STRING, "constrained-baseline", NULL);
}
} else {
/* No profile-level-id means baseline or unrestricted */
gst_structure_set (new_s,
"profile", G_TYPE_STRING, "constrained-baseline", NULL);
append_unrestricted = TRUE;
}
caps = gst_caps_merge_structure (caps, new_s);
}
if (append_unrestricted) {
caps =
gst_caps_merge_structure (caps, gst_structure_new ("video/x-h264", NULL,
NULL));
}
icaps = gst_caps_intersect (caps, template_caps);
gst_caps_unref (caps);
caps = icaps;
done:
if (filter) {
GST_DEBUG_OBJECT (payload, "Intersect %" GST_PTR_FORMAT " and filter %"
GST_PTR_FORMAT, caps, filter);
icaps = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = icaps;
}
gst_caps_unref (template_caps);
gst_caps_unref (allowed_caps);
GST_LOG_OBJECT (payload, "returning caps %" GST_PTR_FORMAT, caps);
return caps;
}
static gboolean
gst_rtp_h264_pay_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
GstRtpH264Pay *rtph264pay = GST_RTP_H264_PAY (parent);
if (GST_QUERY_TYPE (query) == GST_QUERY_LATENCY) {
gboolean retval;
gboolean live;
GstClockTime min_latency, max_latency;
retval = gst_pad_query_default (pad, parent, query);
if (!retval)
return retval;
if (rtph264pay->stream_format == GST_H264_STREAM_FORMAT_UNKNOWN ||
rtph264pay->alignment == GST_H264_ALIGNMENT_UNKNOWN)
return FALSE;
gst_query_parse_latency (query, &live, &min_latency, &max_latency);
if (rtph264pay->aggregate_mode == GST_RTP_H264_AGGREGATE_MAX_STAP &&
rtph264pay->alignment != GST_H264_ALIGNMENT_AU && rtph264pay->fps_num) {
GstClockTime one_frame = gst_util_uint64_scale_int (GST_SECOND,
rtph264pay->fps_denum, rtph264pay->fps_num);
min_latency += one_frame;
max_latency += one_frame;
gst_query_set_latency (query, live, min_latency, max_latency);
}
return TRUE;
}
return gst_pad_query_default (pad, parent, query);
}
/* 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 (GstRTPBasePayload * basepayload)
{
GstRtpH264Pay *payloader = GST_RTP_H264_PAY (basepayload);
gchar *profile;
gchar *set;
GString *sprops;
guint count;
gboolean res;
GstMapInfo map;
guint i;
sprops = g_string_new ("");
count = 0;
/* build the sprop-parameter-sets */
for (i = 0; i < payloader->sps->len; i++) {
GstBuffer *sps_buf =
GST_BUFFER_CAST (g_ptr_array_index (payloader->sps, i));
gst_buffer_map (sps_buf, &map, GST_MAP_READ);
set = g_base64_encode (map.data, map.size);
gst_buffer_unmap (sps_buf, &map);
g_string_append_printf (sprops, "%s%s", count ? "," : "", set);
g_free (set);
count++;
}
for (i = 0; i < payloader->pps->len; i++) {
GstBuffer *pps_buf =
GST_BUFFER_CAST (g_ptr_array_index (payloader->pps, i));
gst_buffer_map (pps_buf, &map, GST_MAP_READ);
set = g_base64_encode (map.data, map.size);
gst_buffer_unmap (pps_buf, &map);
g_string_append_printf (sprops, "%s%s", count ? "," : "", set);
g_free (set);
count++;
}
if (G_LIKELY (count)) {
if (payloader->profile != 0) {
/* profile is 24 bit. Force it to respect the limit */
profile = g_strdup_printf ("%06x", payloader->profile & 0xffffff);
/* combine into output caps */
res = gst_rtp_base_payload_set_outcaps (basepayload,
"packetization-mode", G_TYPE_STRING, "1",
"profile-level-id", G_TYPE_STRING, profile,
"sprop-parameter-sets", G_TYPE_STRING, sprops->str, NULL);
g_free (profile);
} else {
res = gst_rtp_base_payload_set_outcaps (basepayload,
"packetization-mode", G_TYPE_STRING, "1",
"sprop-parameter-sets", G_TYPE_STRING, sprops->str, NULL);
}
} else {
res = gst_rtp_base_payload_set_outcaps (basepayload, NULL);
}
g_string_free (sprops, TRUE);
return res;
}
static gboolean
gst_rtp_h264_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
{
GstRtpH264Pay *rtph264pay;
GstStructure *str;
const GValue *value;
GstMapInfo map;
guint8 *data;
gsize size;
GstBuffer *buffer;
const gchar *alignment, *stream_format;
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_rtp_base_payload_set_options (basepayload, "video", TRUE, "H264", 90000);
rtph264pay->alignment = GST_H264_ALIGNMENT_UNKNOWN;
alignment = gst_structure_get_string (str, "alignment");
if (alignment) {
if (g_str_equal (alignment, "au"))
rtph264pay->alignment = GST_H264_ALIGNMENT_AU;
if (g_str_equal (alignment, "nal"))
rtph264pay->alignment = GST_H264_ALIGNMENT_NAL;
}
rtph264pay->stream_format = GST_H264_STREAM_FORMAT_UNKNOWN;
stream_format = gst_structure_get_string (str, "stream-format");
if (stream_format) {
if (g_str_equal (stream_format, "avc"))
rtph264pay->stream_format = GST_H264_STREAM_FORMAT_AVC;
if (g_str_equal (stream_format, "byte-stream"))
rtph264pay->stream_format = GST_H264_STREAM_FORMAT_BYTESTREAM;
}
if (!gst_structure_get_fraction (str, "framerate", &rtph264pay->fps_num,
&rtph264pay->fps_denum))
rtph264pay->fps_num = rtph264pay->fps_denum = 0;
/* packetized AVC video has a codec_data */
if ((value = gst_structure_get_value (str, "codec_data"))) {
guint num_sps, num_pps;
gint i, nal_size;
GST_DEBUG_OBJECT (rtph264pay, "have packetized h264");
buffer = gst_value_get_buffer (value);
gst_buffer_map (buffer, &map, GST_MAP_READ);
data = map.data;
size = map.size;
/* 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);
gst_buffer_fill (sps_buf, 0, data, nal_size);
gst_rtp_h264_add_sps_pps (GST_ELEMENT (rtph264pay), rtph264pay->sps,
rtph264pay->pps, 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);
gst_buffer_fill (pps_buf, 0, data, nal_size);
gst_rtp_h264_add_sps_pps (GST_ELEMENT (rtph264pay), rtph264pay->sps,
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))
goto set_sps_pps_failed;
gst_buffer_unmap (buffer, &map);
} else {
GST_DEBUG_OBJECT (rtph264pay, "have bytestream h264");
}
return TRUE;
avcc_too_small:
{
GST_ERROR_OBJECT (rtph264pay, "avcC size %" G_GSIZE_FORMAT " < 7", size);
goto error;
}
wrong_version:
{
GST_ERROR_OBJECT (rtph264pay, "wrong avcC version");
goto error;
}
avcc_error:
{
GST_ERROR_OBJECT (rtph264pay, "avcC too small ");
goto error;
}
set_sps_pps_failed:
{
GST_ERROR_OBJECT (rtph264pay, "failed to set sps/pps");
goto error;
}
error:
{
gst_buffer_unmap (buffer, &map);
return FALSE;
}
}
static void
gst_rtp_h264_pay_parse_sprop_parameter_sets (GstRtpH264Pay * rtph264pay)
{
const gchar *ps;
gchar **params;
guint len;
gint i;
GstBuffer *buf;
ps = rtph264pay->sprop_parameter_sets;
if (ps == NULL)
return;
gst_rtp_h264_pay_clear_sps_pps (rtph264pay);
params = g_strsplit (ps, ",", 0);
len = g_strv_length (params);
GST_DEBUG_OBJECT (rtph264pay, "we have %d params", len);
for (i = 0; params[i]; i++) {
gsize nal_len;
GstMapInfo map;
guint8 *nalp;
guint save = 0;
gint state = 0;
nal_len = strlen (params[i]);
buf = gst_buffer_new_and_alloc (nal_len);
gst_buffer_map (buf, &map, GST_MAP_WRITE);
nalp = map.data;
nal_len = g_base64_decode_step (params[i], nal_len, nalp, &state, &save);
gst_buffer_unmap (buf, &map);
gst_buffer_resize (buf, 0, nal_len);
if (!nal_len) {
gst_buffer_unref (buf);
continue;
}
gst_rtp_h264_add_sps_pps (GST_ELEMENT (rtph264pay), rtph264pay->sps,
rtph264pay->pps, buf);
}
g_strfreev (params);
}
static guint
next_start_code (const 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 3th byte */
guint offset = 2;
while (offset < size) {
if (1 == data[offset]) {
unsigned int shift = offset;
if (0 == data[--shift]) {
if (0 == data[--shift]) {
return shift;
}
}
/* The jump is always 3 because of the 1 previously matched.
* All the 0's must be after this '1' matched at offset */
offset += 3;
} else if (0 == data[offset]) {
/* maybe next byte is 1? */
offset++;
} else {
/* can jump 3 bytes forward */
offset += 3;
}
/* at each iteration, we rescan in a backward manner until
* we match 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,
const guint8 * data, guint size, GstClockTime dts, GstClockTime pts)
{
guint8 header, type;
gboolean updated;
/* default is no update */
updated = FALSE;
GST_DEBUG ("NAL payload len=%u", size);
header = data[0];
type = header & 0x1f;
/* We 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 || PPS_TYPE_ID == type) {
GstBuffer *nal;
/* trailing 0x0 are not part of the SPS/PPS */
while (size > 0 && data[size - 1] == 0x0)
size--;
/* encode the entire SPS NAL in base64 */
GST_DEBUG ("Found %s %x %x %x Len=%u", type == SPS_TYPE_ID ? "SPS" : "PPS",
(header >> 7), (header >> 5) & 3, type, size);
nal = gst_buffer_new_allocate (NULL, size, NULL);
gst_buffer_fill (nal, 0, data, size);
updated = gst_rtp_h264_add_sps_pps (GST_ELEMENT (payloader),
payloader->sps, payloader->pps, nal);
/* remember when we last saw SPS */
if (pts != -1)
payloader->last_spspps =
gst_segment_to_running_time (&GST_RTP_BASE_PAYLOAD_CAST
(payloader)->segment, GST_FORMAT_TIME, pts);
} else {
GST_DEBUG ("NAL: %x %x %x Len = %u", (header >> 7),
(header >> 5) & 3, type, size);
}
return updated;
}
static GstFlowReturn
gst_rtp_h264_pay_payload_nal (GstRTPBasePayload * basepayload,
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
gboolean delta_unit, gboolean discont);
static GstFlowReturn
gst_rtp_h264_pay_payload_nal_single (GstRTPBasePayload * basepayload,
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
gboolean delta_unit, gboolean discont);
static GstFlowReturn
gst_rtp_h264_pay_payload_nal_fragment (GstRTPBasePayload * basepayload,
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
gboolean delta_unit, gboolean discont, guint8 nal_header);
static GstFlowReturn
gst_rtp_h264_pay_payload_nal_bundle (GstRTPBasePayload * basepayload,
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
gboolean delta_unit, gboolean discont, guint8 nal_header);
static GstFlowReturn
gst_rtp_h264_pay_send_sps_pps (GstRTPBasePayload * basepayload,
GstClockTime dts, GstClockTime pts, gboolean delta_unit, gboolean discont)
{
GstRtpH264Pay *rtph264pay = GST_RTP_H264_PAY (basepayload);
GstFlowReturn ret = GST_FLOW_OK;
gboolean sent_all_sps_pps = TRUE;
guint i;
for (i = 0; i < rtph264pay->sps->len; i++) {
GstBuffer *sps_buf =
GST_BUFFER_CAST (g_ptr_array_index (rtph264pay->sps, i));
GST_DEBUG_OBJECT (rtph264pay, "inserting SPS in the stream");
/* resend SPS */
ret = gst_rtp_h264_pay_payload_nal (basepayload, gst_buffer_ref (sps_buf),
dts, pts, FALSE, delta_unit, discont);
/* Not critical here; but throw a warning */
if (ret != GST_FLOW_OK) {
sent_all_sps_pps = FALSE;
GST_WARNING_OBJECT (basepayload, "Problem pushing SPS");
}
}
for (i = 0; i < rtph264pay->pps->len; i++) {
GstBuffer *pps_buf =
GST_BUFFER_CAST (g_ptr_array_index (rtph264pay->pps, i));
GST_DEBUG_OBJECT (rtph264pay, "inserting PPS in the stream");
/* resend PPS */
ret = gst_rtp_h264_pay_payload_nal (basepayload, gst_buffer_ref (pps_buf),
dts, pts, FALSE, TRUE, FALSE);
/* Not critical here; but throw a warning */
if (ret != GST_FLOW_OK) {
sent_all_sps_pps = FALSE;
GST_WARNING_OBJECT (basepayload, "Problem pushing PPS");
}
}
if (pts != -1 && sent_all_sps_pps)
rtph264pay->last_spspps =
gst_segment_to_running_time (&basepayload->segment, GST_FORMAT_TIME,
pts);
return ret;
}
/* @delta_unit: if %FALSE the first packet sent won't have the
* GST_BUFFER_FLAG_DELTA_UNIT flag.
* @discont: if %TRUE the first packet sent will have the
* GST_BUFFER_FLAG_DISCONT flag.
*/
static GstFlowReturn
gst_rtp_h264_pay_payload_nal (GstRTPBasePayload * basepayload,
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
gboolean delta_unit, gboolean discont)
{
GstRtpH264Pay *rtph264pay;
guint8 nal_header, nal_type;
gboolean send_spspps;
guint size;
rtph264pay = GST_RTP_H264_PAY (basepayload);
size = gst_buffer_get_size (paybuf);
gst_buffer_extract (paybuf, 0, &nal_header, 1);
nal_type = nal_header & 0x1f;
/* These payload type are reserved for STAP-A, STAP-B, MTAP16, and MTAP24
* as internally used NAL types */
switch (nal_type) {
case 24:
case 25:
case 26:
case 27:
GST_WARNING_OBJECT (rtph264pay, "Ignoring reserved NAL TYPE=%d",
nal_type);
gst_buffer_unref (paybuf);
return GST_FLOW_OK;
default:
break;
}
GST_DEBUG_OBJECT (rtph264pay,
"payloading NAL Unit: datasize=%u type=%d pts=%" GST_TIME_FORMAT,
size, nal_type, GST_TIME_ARGS (pts));
/* should set src caps before pushing stuff,
* and if we did not see enough SPS/PPS, that may not be the case */
if (G_UNLIKELY (!gst_pad_has_current_caps (GST_RTP_BASE_PAYLOAD_SRCPAD
(basepayload))))
gst_rtp_h264_pay_set_sps_pps (basepayload);
send_spspps = FALSE;
/* check if we need to emit an SPS/PPS now */
if (nal_type == IDR_TYPE_ID && rtph264pay->spspps_interval > 0) {
if (rtph264pay->last_spspps != -1) {
guint64 diff;
GstClockTime running_time =
gst_segment_to_running_time (&basepayload->segment, GST_FORMAT_TIME,
pts);
GST_LOG_OBJECT (rtph264pay,
"now %" GST_TIME_FORMAT ", last SPS/PPS %" GST_TIME_FORMAT,
GST_TIME_ARGS (running_time),
GST_TIME_ARGS (rtph264pay->last_spspps));
/* calculate diff between last SPS/PPS in milliseconds */
if (running_time > rtph264pay->last_spspps)
diff = running_time - rtph264pay->last_spspps;
else
diff = 0;
GST_DEBUG_OBJECT (rtph264pay,
"interval since last SPS/PPS %" GST_TIME_FORMAT,
GST_TIME_ARGS (diff));
/* bigger than interval, queue SPS/PPS */
if (GST_TIME_AS_SECONDS (diff) >= rtph264pay->spspps_interval) {
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;
}
} else if (nal_type == IDR_TYPE_ID && rtph264pay->spspps_interval == -1) {
GST_DEBUG_OBJECT (rtph264pay, "sending SPS/PPS before current IDR frame");
/* send SPS/PPS before every IDR frame */
send_spspps = TRUE;
}
if (send_spspps || rtph264pay->send_spspps) {
/* we need to send SPS/PPS now first. FIXME, don't use the pts for
* checking when we need to send SPS/PPS but convert to running_time first. */
GstFlowReturn ret;
rtph264pay->send_spspps = FALSE;
ret = gst_rtp_h264_pay_send_sps_pps (basepayload, dts, pts, delta_unit,
discont);
if (ret != GST_FLOW_OK) {
gst_buffer_unref (paybuf);
return ret;
}
delta_unit = TRUE;
discont = FALSE;
}
if (rtph264pay->aggregate_mode != GST_RTP_H264_AGGREGATE_NONE)
return gst_rtp_h264_pay_payload_nal_bundle (basepayload, paybuf, dts, pts,
end_of_au, delta_unit, discont, nal_header);
return gst_rtp_h264_pay_payload_nal_fragment (basepayload, paybuf, dts, pts,
end_of_au, delta_unit, discont, nal_header);
}
static GstFlowReturn
gst_rtp_h264_pay_payload_nal_fragment (GstRTPBasePayload * basepayload,
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
gboolean delta_unit, gboolean discont, guint8 nal_header)
{
GstRtpH264Pay *rtph264pay;
guint mtu, size, max_fragment_size, max_fragments, ii, pos;
GstBuffer *outbuf;
guint8 *payload;
GstBufferList *list = NULL;
GstRTPBuffer rtp = { NULL };
rtph264pay = GST_RTP_H264_PAY (basepayload);
mtu = GST_RTP_BASE_PAYLOAD_MTU (rtph264pay);
size = gst_buffer_get_size (paybuf);
if (gst_rtp_buffer_calc_packet_len (size, 0, 0) <= mtu) {
/* We don't need to fragment this packet */
GST_DEBUG_OBJECT (rtph264pay,
"sending NAL Unit: datasize=%u mtu=%u", size, mtu);
return gst_rtp_h264_pay_payload_nal_single (basepayload, paybuf, dts, pts,
end_of_au, delta_unit, discont);
}
GST_DEBUG_OBJECT (basepayload,
"using FU-A fragmentation for NAL Unit: datasize=%u mtu=%u", size, mtu);
/* We keep 2 bytes for FU indicator and FU Header */
max_fragment_size = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0);
max_fragments = (size + max_fragment_size - 2) / max_fragment_size;
list = gst_buffer_list_new_sized (max_fragments);
/* Start at the NALU payload */
for (pos = 1, ii = 0; pos < size; pos += max_fragment_size, ii++) {
guint remaining, fragment_size;
gboolean first_fragment, last_fragment;
remaining = size - pos;
fragment_size = MIN (remaining, max_fragment_size);
first_fragment = (pos == 1);
last_fragment = (remaining <= max_fragment_size);
GST_DEBUG_OBJECT (basepayload,
"creating FU-A packet %u/%u, size %u",
ii + 1, max_fragments, fragment_size);
/* use buffer lists
* create buffer without payload containing only the RTP header
* (memory block at index 0) */
outbuf = gst_rtp_buffer_new_allocate (2, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
GST_BUFFER_DTS (outbuf) = dts;
GST_BUFFER_PTS (outbuf) = pts;
payload = gst_rtp_buffer_get_payload (&rtp);
/* If it's the last fragment and the end of this au, mark the end of
* slice */
gst_rtp_buffer_set_marker (&rtp, last_fragment && end_of_au);
/* FU indicator */
payload[0] = (nal_header & 0x60) | FU_A_TYPE_ID;
/* FU Header */
payload[1] = (first_fragment << 7) | (last_fragment << 6) |
(nal_header & 0x1f);
gst_rtp_buffer_unmap (&rtp);
/* insert payload memory block */
gst_rtp_copy_video_meta (rtph264pay, outbuf, paybuf);
gst_buffer_copy_into (outbuf, paybuf, GST_BUFFER_COPY_MEMORY, pos,
fragment_size);
if (!delta_unit)
/* Only the first packet sent should not have the flag */
delta_unit = TRUE;
else
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DELTA_UNIT);
if (discont) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
/* Only the first packet sent should have the flag */
discont = FALSE;
}
/* add the buffer to the buffer list */
gst_buffer_list_add (list, outbuf);
}
GST_DEBUG_OBJECT (rtph264pay,
"sending FU-A fragments: n=%u datasize=%u mtu=%u", ii, size, mtu);
gst_buffer_unref (paybuf);
return gst_rtp_base_payload_push_list (basepayload, list);
}
static GstFlowReturn
gst_rtp_h264_pay_payload_nal_single (GstRTPBasePayload * basepayload,
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
gboolean delta_unit, gboolean discont)
{
GstRtpH264Pay *rtph264pay;
GstBuffer *outbuf;
GstRTPBuffer rtp = { NULL };
rtph264pay = GST_RTP_H264_PAY (basepayload);
/* create buffer without payload containing only the RTP header
* (memory block at index 0) */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
/* Mark the end of a frame */
gst_rtp_buffer_set_marker (&rtp, end_of_au);
/* timestamp the outbuffer */
GST_BUFFER_PTS (outbuf) = pts;
GST_BUFFER_DTS (outbuf) = dts;
if (delta_unit)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DELTA_UNIT);
if (discont)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
gst_rtp_buffer_unmap (&rtp);
/* insert payload memory block */
gst_rtp_copy_video_meta (rtph264pay, outbuf, paybuf);
outbuf = gst_buffer_append (outbuf, paybuf);
/* push the buffer to the next element */
return gst_rtp_base_payload_push (basepayload, outbuf);
}
static void
gst_rtp_h264_pay_reset_bundle (GstRtpH264Pay * rtph264pay)
{
g_clear_pointer (&rtph264pay->bundle, gst_buffer_list_unref);
rtph264pay->bundle_size = 0;
rtph264pay->bundle_contains_vcl = FALSE;
}
static GstFlowReturn
gst_rtp_h264_pay_send_bundle (GstRtpH264Pay * rtph264pay, gboolean end_of_au)
{
GstRTPBasePayload *basepayload;
GstBufferList *bundle;
guint length, bundle_size;
GstBuffer *first, *outbuf;
GstClockTime dts, pts;
gboolean delta, discont;
bundle_size = rtph264pay->bundle_size;
if (bundle_size == 0) {
GST_DEBUG_OBJECT (rtph264pay, "no bundle, nothing to send");
return GST_FLOW_OK;
}
basepayload = GST_RTP_BASE_PAYLOAD (rtph264pay);
bundle = rtph264pay->bundle;
length = gst_buffer_list_length (bundle);
first = gst_buffer_list_get (bundle, 0);
dts = GST_BUFFER_DTS (first);
pts = GST_BUFFER_PTS (first);
delta = GST_BUFFER_FLAG_IS_SET (first, GST_BUFFER_FLAG_DELTA_UNIT);
discont = GST_BUFFER_FLAG_IS_SET (first, GST_BUFFER_FLAG_DISCONT);
if (length == 1) {
/* Push unaggregated NALU */
outbuf = gst_buffer_ref (first);
GST_DEBUG_OBJECT (rtph264pay,
"sending NAL Unit unaggregated: datasize=%u", bundle_size - 2);
} else {
guint8 stap_header;
guint i;
outbuf = gst_buffer_new_allocate (NULL, sizeof stap_header, NULL);
stap_header = STAP_A_TYPE_ID;
for (i = 0; i < length; i++) {
GstBuffer *buf = gst_buffer_list_get (bundle, i);
guint8 nal_header;
GstMemory *size_header;
GstMapInfo map;
gst_buffer_extract (buf, 0, &nal_header, sizeof nal_header);
/* Propagate F bit */
if ((nal_header & 0x80))
stap_header |= 0x80;
/* Select highest nal_ref_idc */
if ((nal_header & 0x60) > (stap_header & 0x60))
stap_header = (stap_header & 0x9f) | (nal_header & 0x60);
/* append NALU size */
size_header = gst_allocator_alloc (NULL, 2, NULL);
gst_memory_map (size_header, &map, GST_MAP_WRITE);
GST_WRITE_UINT16_BE (map.data, gst_buffer_get_size (buf));
gst_memory_unmap (size_header, &map);
gst_buffer_append_memory (outbuf, size_header);
/* append NALU data */
outbuf = gst_buffer_append (outbuf, gst_buffer_ref (buf));
}
gst_buffer_fill (outbuf, 0, &stap_header, sizeof stap_header);
GST_DEBUG_OBJECT (rtph264pay,
"sending STAP-A bundle: n=%u header=%02x datasize=%u",
length, stap_header, bundle_size);
}
gst_rtp_h264_pay_reset_bundle (rtph264pay);
return gst_rtp_h264_pay_payload_nal_single (basepayload, outbuf, dts, pts,
end_of_au, delta, discont);
}
static gboolean
gst_rtp_h264_pay_payload_nal_bundle (GstRTPBasePayload * basepayload,
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
gboolean delta_unit, gboolean discont, guint8 nal_header)
{
GstRtpH264Pay *rtph264pay;
GstFlowReturn ret;
guint mtu, pay_size, bundle_size;
GstBufferList *bundle;
guint8 nal_type;
gboolean start_of_au;
rtph264pay = GST_RTP_H264_PAY (basepayload);
nal_type = nal_header & 0x1f;
mtu = GST_RTP_BASE_PAYLOAD_MTU (rtph264pay);
pay_size = 2 + gst_buffer_get_size (paybuf);
bundle = rtph264pay->bundle;
start_of_au = FALSE;
if (bundle) {
GstBuffer *first = gst_buffer_list_get (bundle, 0);
if (nal_type == AUD_TYPE_ID) {
GST_DEBUG_OBJECT (rtph264pay, "found access delimiter");
start_of_au = TRUE;
} else if (discont) {
GST_DEBUG_OBJECT (rtph264pay, "found discont");
start_of_au = TRUE;
} else if (GST_BUFFER_PTS (first) != pts || GST_BUFFER_DTS (first) != dts) {
GST_DEBUG_OBJECT (rtph264pay, "found timestamp mismatch");
start_of_au = TRUE;
}
}
if (start_of_au) {
GST_DEBUG_OBJECT (rtph264pay, "sending bundle before start of AU");
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, TRUE);
if (ret != GST_FLOW_OK)
goto out;
bundle = NULL;
}
bundle_size = 1 + pay_size;
if (gst_rtp_buffer_calc_packet_len (bundle_size, 0, 0) > mtu) {
GST_DEBUG_OBJECT (rtph264pay, "NAL Unit cannot fit in a bundle");
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, FALSE);
if (ret != GST_FLOW_OK)
goto out;
return gst_rtp_h264_pay_payload_nal_fragment (basepayload, paybuf, dts, pts,
end_of_au, delta_unit, discont, nal_header);
}
bundle_size = rtph264pay->bundle_size + pay_size;
if (gst_rtp_buffer_calc_packet_len (bundle_size, 0, 0) > mtu) {
GST_DEBUG_OBJECT (rtph264pay,
"bundle overflows, sending: bundlesize=%u datasize=2+%u mtu=%u",
rtph264pay->bundle_size, pay_size - 2, mtu);
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, FALSE);
if (ret != GST_FLOW_OK)
goto out;
bundle = NULL;
}
if (!bundle) {
GST_DEBUG_OBJECT (rtph264pay, "creating new STAP-A aggregate");
bundle = rtph264pay->bundle = gst_buffer_list_new ();
bundle_size = rtph264pay->bundle_size = 1;
rtph264pay->bundle_contains_vcl = FALSE;
}
GST_DEBUG_OBJECT (rtph264pay,
"bundling NAL Unit: bundlesize=%u datasize=2+%u mtu=%u",
rtph264pay->bundle_size, pay_size - 2, mtu);
paybuf = gst_buffer_make_writable (paybuf);
GST_BUFFER_PTS (paybuf) = pts;
GST_BUFFER_DTS (paybuf) = dts;
if (delta_unit)
GST_BUFFER_FLAG_SET (paybuf, GST_BUFFER_FLAG_DELTA_UNIT);
else
GST_BUFFER_FLAG_UNSET (paybuf, GST_BUFFER_FLAG_DELTA_UNIT);
if (discont)
GST_BUFFER_FLAG_SET (paybuf, GST_BUFFER_FLAG_DISCONT);
else
GST_BUFFER_FLAG_UNSET (paybuf, GST_BUFFER_FLAG_DISCONT);
gst_buffer_list_add (bundle, gst_buffer_ref (paybuf));
rtph264pay->bundle_size += pay_size;
ret = GST_FLOW_OK;
if ((nal_type >= 1 && nal_type <= 5) || nal_type == 14 ||
(nal_type >= 20 && nal_type <= 23))
rtph264pay->bundle_contains_vcl = TRUE;
if (end_of_au) {
GST_DEBUG_OBJECT (rtph264pay, "sending bundle at end of AU");
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, TRUE);
}
out:
gst_buffer_unref (paybuf);
return ret;
}
static GstFlowReturn
gst_rtp_h264_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpH264Pay *rtph264pay;
GstFlowReturn ret;
gsize size;
guint nal_len, i;
const guint8 *data;
GstClockTime dts, pts;
GArray *nal_queue;
gboolean avc;
GstBuffer *paybuf = NULL;
gsize skip;
gboolean delayed_not_delta_unit = FALSE;
gboolean delayed_discont = FALSE;
gboolean marker = FALSE;
gboolean draining = (buffer == NULL);
rtph264pay = GST_RTP_H264_PAY (basepayload);
/* the input buffer contains one or more NAL units */
avc = rtph264pay->stream_format == GST_H264_STREAM_FORMAT_AVC;
if (avc) {
/* In AVC mode, there is no adapter, so nothing to drain */
if (draining)
return GST_FLOW_OK;
} else {
if (buffer) {
if (!GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DELTA_UNIT)) {
if (gst_adapter_available (rtph264pay->adapter) == 0)
rtph264pay->delta_unit = FALSE;
else
/* This buffer contains a key frame but the adapter isn't empty. So
* we'll purge it first by sending a first packet and then the second
* one won't have the DELTA_UNIT flag. */
delayed_not_delta_unit = TRUE;
}
if (GST_BUFFER_IS_DISCONT (buffer)) {
if (gst_adapter_available (rtph264pay->adapter) == 0)
rtph264pay->discont = TRUE;
else
/* This buffer has the DISCONT flag but the adapter isn't empty. So
* we'll purge it first by sending a first packet and then the second
* one will have the DISCONT flag set. */
delayed_discont = TRUE;
}
marker = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_MARKER);
gst_adapter_push (rtph264pay->adapter, buffer);
buffer = NULL;
}
/* We want to use the first TS used to construct the following NAL */
dts = gst_adapter_prev_dts (rtph264pay->adapter, NULL);
pts = gst_adapter_prev_pts (rtph264pay->adapter, NULL);
size = gst_adapter_available (rtph264pay->adapter);
/* Nothing to do here if the adapter is empty, e.g. on EOS */
if (size == 0)
return GST_FLOW_OK;
data = gst_adapter_map (rtph264pay->adapter, size);
GST_DEBUG_OBJECT (basepayload, "got %" G_GSIZE_FORMAT " bytes", size);
}
ret = GST_FLOW_OK;
/* now loop over all NAL units and put them in a packet */
if (avc) {
GstBufferMemoryMap memory;
gsize remaining_buffer_size;
guint nal_length_size;
gsize offset = 0;
gst_buffer_memory_map (buffer, &memory);
remaining_buffer_size = gst_buffer_get_size (buffer);
pts = GST_BUFFER_PTS (buffer);
dts = GST_BUFFER_DTS (buffer);
rtph264pay->delta_unit = GST_BUFFER_FLAG_IS_SET (buffer,
GST_BUFFER_FLAG_DELTA_UNIT);
rtph264pay->discont = GST_BUFFER_IS_DISCONT (buffer);
marker = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_MARKER);
GST_DEBUG_OBJECT (basepayload, "got %" G_GSIZE_FORMAT " bytes",
remaining_buffer_size);
nal_length_size = rtph264pay->nal_length_size;
while (remaining_buffer_size > nal_length_size) {
gint i;
gboolean end_of_au = FALSE;
nal_len = 0;
for (i = 0; i < nal_length_size; i++) {
nal_len = (nal_len << 8) + *memory.data;
if (!gst_buffer_memory_advance_bytes (&memory, 1))
break;
}
offset += nal_length_size;
remaining_buffer_size -= nal_length_size;
if (remaining_buffer_size >= nal_len) {
GST_DEBUG_OBJECT (basepayload, "got NAL of size %u", nal_len);
} else {
nal_len = remaining_buffer_size;
GST_DEBUG_OBJECT (basepayload, "got incomplete NAL of size %u",
nal_len);
}
/* If we're at the end of the buffer, then we're at the end of the
* access unit
*/
if (remaining_buffer_size - nal_len <= nal_length_size) {
if (rtph264pay->alignment == GST_H264_ALIGNMENT_AU || marker)
end_of_au = TRUE;
}
paybuf = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, offset,
nal_len);
ret =
gst_rtp_h264_pay_payload_nal (basepayload, paybuf, dts, pts,
end_of_au, rtph264pay->delta_unit, rtph264pay->discont);
if (!rtph264pay->delta_unit)
/* Only the first outgoing packet doesn't have the DELTA_UNIT flag */
rtph264pay->delta_unit = TRUE;
if (rtph264pay->discont)
/* Only the first outgoing packet have the DISCONT flag */
rtph264pay->discont = FALSE;
if (ret != GST_FLOW_OK)
break;
/* Skip current nal. If it is split over multiple GstMemory
* advance_bytes () will switch to the correct GstMemory. The payloader
* does not access those bytes directly but uses gst_buffer_copy_region ()
* to create a sub-buffer referencing the nal instead */
if (!gst_buffer_memory_advance_bytes (&memory, nal_len))
break;
offset += nal_len;
remaining_buffer_size -= nal_len;
}
gst_buffer_memory_unmap (&memory);
gst_buffer_unref (buffer);
} 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_queue = rtph264pay->queue;
skip = next;
/* 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 %" G_GSIZE_FORMAT, next, size);
/* first pass to locate NALs and parse SPS/PPS */
while (size > 4) {
/* skip start code */
data += 3;
size -= 3;
/* 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 or au aligned input needs no delaying until next time */
if (next == size && !draining &&
rtph264pay->alignment == GST_H264_ALIGNMENT_UNKNOWN) {
/* Didn't find the start of next NAL and it's not EOS,
* handle it next time */
break;
}
/* 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_rtp_base_payload_set_outcaps (basepayload,
"sprop-parameter-sets", G_TYPE_STRING,
rtph264pay->sprop_parameter_sets, NULL))
goto caps_rejected;
/* parse SPS and PPS from provided parameter set (for insertion) */
gst_rtp_h264_pay_parse_sprop_parameter_sets (rtph264pay);
rtph264pay->update_caps = FALSE;
GST_DEBUG ("outcaps update: 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, dts, pts)
|| 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 */
if (nal_queue->len != 0)
gst_adapter_flush (rtph264pay->adapter, skip);
for (i = 0; i < nal_queue->len; i++) {
guint size;
gboolean end_of_au = FALSE;
nal_len = g_array_index (nal_queue, guint, i);
/* skip start code */
gst_adapter_flush (rtph264pay->adapter, 3);
/* Trim the end unless we're the last NAL in the stream.
* In case we're not at the end of the buffer we know the next block
* starts with 0x000001 so all the 0x00 bytes at the end of this one are
* trailing 0x0 that can be discarded */
size = nal_len;
data = gst_adapter_map (rtph264pay->adapter, size);
if (i + 1 != nal_queue->len || !draining)
for (; size > 1 && data[size - 1] == 0x0; size--)
/* skip */ ;
/* If it's the last nal unit we have in non-bytestream mode, we can
* assume it's the end of an access-unit
*
* FIXME: We need to wait until the next packet or EOS to
* actually payload the NAL so we can know if the current NAL is
* the last one of an access unit or not if we are in bytestream mode
*/
if (i == nal_queue->len - 1) {
if (rtph264pay->alignment == GST_H264_ALIGNMENT_AU ||
marker || draining)
end_of_au = TRUE;
}
paybuf = gst_adapter_take_buffer (rtph264pay->adapter, size);
g_assert (paybuf);
/* put the data in one or more RTP packets */
ret =
gst_rtp_h264_pay_payload_nal (basepayload, paybuf, dts, pts,
end_of_au, rtph264pay->delta_unit, rtph264pay->discont);
if (delayed_not_delta_unit) {
rtph264pay->delta_unit = FALSE;
delayed_not_delta_unit = FALSE;
} else {
/* Only the first outgoing packet doesn't have the DELTA_UNIT flag */
rtph264pay->delta_unit = TRUE;
}
if (delayed_discont) {
rtph264pay->discont = TRUE;
delayed_discont = FALSE;
} else {
/* Only the first outgoing packet have the DISCONT flag */
rtph264pay->discont = FALSE;
}
if (ret != GST_FLOW_OK) {
break;
}
/* move to next NAL packet */
/* Skips the trailing zeros */
gst_adapter_flush (rtph264pay->adapter, nal_len - size);
}
g_array_set_size (nal_queue, 0);
}
if (ret == GST_FLOW_OK && rtph264pay->bundle_size > 0 &&
rtph264pay->aggregate_mode == GST_RTP_H264_AGGREGATE_ZERO_LATENCY &&
rtph264pay->bundle_contains_vcl) {
GST_DEBUG_OBJECT (rtph264pay, "sending bundle at end incoming packet");
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, FALSE);
}
done:
if (!avc) {
gst_adapter_unmap (rtph264pay->adapter);
}
return ret;
caps_rejected:
{
GST_WARNING_OBJECT (basepayload, "Could not set outcaps");
g_array_set_size (nal_queue, 0);
ret = GST_FLOW_NOT_NEGOTIATED;
goto done;
}
}
static gboolean
gst_rtp_h264_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
{
gboolean res;
const GstStructure *s;
GstRtpH264Pay *rtph264pay = GST_RTP_H264_PAY (payload);
GstFlowReturn ret = GST_FLOW_OK;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
gst_adapter_clear (rtph264pay->adapter);
gst_rtp_h264_pay_reset_bundle (rtph264pay);
break;
case GST_EVENT_CUSTOM_DOWNSTREAM:
s = gst_event_get_structure (event);
if (gst_structure_has_name (s, "GstForceKeyUnit")) {
gboolean resend_codec_data;
if (gst_structure_get_boolean (s, "all-headers",
&resend_codec_data) && resend_codec_data)
rtph264pay->send_spspps = TRUE;
}
break;
case GST_EVENT_EOS:
{
/* call handle_buffer with NULL to flush last NAL from adapter
* in byte-stream mode
*/
gst_rtp_h264_pay_handle_buffer (payload, NULL);
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, TRUE);
break;
}
case GST_EVENT_STREAM_START:
GST_DEBUG_OBJECT (rtph264pay, "New stream detected => Clear SPS and PPS");
gst_rtp_h264_pay_clear_sps_pps (rtph264pay);
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, TRUE);
break;
default:
break;
}
if (ret != GST_FLOW_OK)
return FALSE;
res = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
return res;
}
static GstStateChangeReturn
gst_rtp_h264_pay_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstRtpH264Pay *rtph264pay = GST_RTP_H264_PAY (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
rtph264pay->send_spspps = FALSE;
gst_adapter_clear (rtph264pay->adapter);
gst_rtp_h264_pay_reset_bundle (rtph264pay);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
rtph264pay->last_spspps = -1;
gst_rtp_h264_pay_clear_sps_pps (rtph264pay);
break;
default:
break;
}
return ret;
}
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_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_CONFIG_INTERVAL:
rtph264pay->spspps_interval = g_value_get_int (value);
break;
case PROP_AGGREGATE_MODE:
rtph264pay->aggregate_mode = g_value_get_enum (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_SPROP_PARAMETER_SETS:
g_value_set_string (value, rtph264pay->sprop_parameter_sets);
break;
case PROP_CONFIG_INTERVAL:
g_value_set_int (value, rtph264pay->spspps_interval);
break;
case PROP_AGGREGATE_MODE:
g_value_set_enum (value, rtph264pay->aggregate_mode);
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_SECONDARY, GST_TYPE_RTP_H264_PAY);
}
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