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gstreamer/gst/rtp/gstrtpvorbispay.c
Sebastian Dröge b1089fb520 rtp: Copy metadata in the (de)payloader, but only the relevant ones 
The payloader didn't copy anything so far, the depayloader copied every
possible meta. Let's make it consistent and just copy all metas without
tags or with only the video tag.

https://bugzilla.gnome.org/show_bug.cgi?id=751774
2015-08-11 12:47:23 +02:00

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/* GStreamer
* Copyright (C) <2006> Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "fnv1hash.h"
#include "gstrtpvorbispay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpvorbispay_debug);
#define GST_CAT_DEFAULT (rtpvorbispay_debug)
/* references:
* http://www.rfc-editor.org/rfc/rfc5215.txt
*/
static GstStaticPadTemplate gst_rtp_vorbis_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"VORBIS\""
/* All required parameters
*
* "encoding-params = (string) <num channels>"
* "configuration = (string) ANY"
*/
)
);
static GstStaticPadTemplate gst_rtp_vorbis_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-vorbis")
);
#define DEFAULT_CONFIG_INTERVAL 0
enum
{
PROP_0,
PROP_CONFIG_INTERVAL
};
#define gst_rtp_vorbis_pay_parent_class parent_class
G_DEFINE_TYPE (GstRtpVorbisPay, gst_rtp_vorbis_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static gboolean gst_rtp_vorbis_pay_setcaps (GstRTPBasePayload * basepayload,
GstCaps * caps);
static GstStateChangeReturn gst_rtp_vorbis_pay_change_state (GstElement *
element, GstStateChange transition);
static GstFlowReturn gst_rtp_vorbis_pay_handle_buffer (GstRTPBasePayload * pad,
GstBuffer * buffer);
static gboolean gst_rtp_vorbis_pay_sink_event (GstRTPBasePayload * payload,
GstEvent * event);
static gboolean gst_rtp_vorbis_pay_parse_id (GstRTPBasePayload * basepayload,
guint8 * data, guint size);
static gboolean gst_rtp_vorbis_pay_finish_headers (GstRTPBasePayload *
basepayload);
static void gst_rtp_vorbis_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_vorbis_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void
gst_rtp_vorbis_pay_class_init (GstRtpVorbisPayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gstelement_class->change_state = gst_rtp_vorbis_pay_change_state;
gstrtpbasepayload_class->set_caps = gst_rtp_vorbis_pay_setcaps;
gstrtpbasepayload_class->handle_buffer = gst_rtp_vorbis_pay_handle_buffer;
gstrtpbasepayload_class->sink_event = gst_rtp_vorbis_pay_sink_event;
gobject_class->set_property = gst_rtp_vorbis_pay_set_property;
gobject_class->get_property = gst_rtp_vorbis_pay_get_property;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_vorbis_pay_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_vorbis_pay_sink_template));
gst_element_class_set_static_metadata (gstelement_class,
"RTP Vorbis payloader",
"Codec/Payloader/Network/RTP",
"Payload-encode Vorbis audio into RTP packets (RFC 5215)",
"Wim Taymans <wim.taymans@gmail.com>");
GST_DEBUG_CATEGORY_INIT (rtpvorbispay_debug, "rtpvorbispay", 0,
"Vorbis RTP Payloader");
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CONFIG_INTERVAL,
g_param_spec_uint ("config-interval", "Config Send Interval",
"Send Config Insertion Interval in seconds (configuration headers "
"will be multiplexed in the data stream when detected.) (0 = disabled)",
0, 3600, DEFAULT_CONFIG_INTERVAL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
);
}
static void
gst_rtp_vorbis_pay_init (GstRtpVorbisPay * rtpvorbispay)
{
rtpvorbispay->last_config = GST_CLOCK_TIME_NONE;
}
static void
gst_rtp_vorbis_pay_clear_packet (GstRtpVorbisPay * rtpvorbispay)
{
if (rtpvorbispay->packet)
gst_buffer_unref (rtpvorbispay->packet);
rtpvorbispay->packet = NULL;
g_list_foreach (rtpvorbispay->packet_buffers, (GFunc) gst_mini_object_unref,
NULL);
g_list_free (rtpvorbispay->packet_buffers);
rtpvorbispay->packet_buffers = NULL;
}
static void
gst_rtp_vorbis_pay_cleanup (GstRtpVorbisPay * rtpvorbispay)
{
g_list_foreach (rtpvorbispay->headers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (rtpvorbispay->headers);
rtpvorbispay->headers = NULL;
gst_rtp_vorbis_pay_clear_packet (rtpvorbispay);
if (rtpvorbispay->config_data)
g_free (rtpvorbispay->config_data);
rtpvorbispay->config_data = NULL;
rtpvorbispay->last_config = GST_CLOCK_TIME_NONE;
}
static gboolean
gst_rtp_vorbis_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
{
GstRtpVorbisPay *rtpvorbispay;
GstStructure *s;
const GValue *array;
gint asize, i;
GstBuffer *buf;
GstMapInfo map;
rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
s = gst_caps_get_structure (caps, 0);
rtpvorbispay->need_headers = TRUE;
if ((array = gst_structure_get_value (s, "streamheader")) == NULL)
goto done;
if (G_VALUE_TYPE (array) != GST_TYPE_ARRAY)
goto done;
if ((asize = gst_value_array_get_size (array)) < 3)
goto done;
for (i = 0; i < asize; i++) {
const GValue *value;
value = gst_value_array_get_value (array, i);
if ((buf = gst_value_get_buffer (value)) == NULL)
goto null_buffer;
gst_buffer_map (buf, &map, GST_MAP_READ);
if (map.size < 1)
goto invalid_streamheader;
/* no data packets allowed */
if ((map.data[0] & 1) == 0)
goto invalid_streamheader;
/* we need packets with id 1, 3, 5 */
if (map.data[0] != (i * 2) + 1)
goto invalid_streamheader;
if (i == 0) {
/* identification, we need to parse this in order to get the clock rate. */
if (G_UNLIKELY (!gst_rtp_vorbis_pay_parse_id (basepayload, map.data,
map.size)))
goto parse_id_failed;
}
GST_DEBUG_OBJECT (rtpvorbispay, "collecting header %d", i);
rtpvorbispay->headers =
g_list_append (rtpvorbispay->headers, gst_buffer_ref (buf));
gst_buffer_unmap (buf, &map);
}
if (!gst_rtp_vorbis_pay_finish_headers (basepayload))
goto finish_failed;
done:
return TRUE;
/* ERRORS */
null_buffer:
{
GST_WARNING_OBJECT (rtpvorbispay, "streamheader with null buffer received");
return FALSE;
}
invalid_streamheader:
{
GST_WARNING_OBJECT (rtpvorbispay, "unable to parse initial header");
gst_buffer_unmap (buf, &map);
return FALSE;
}
parse_id_failed:
{
GST_WARNING_OBJECT (rtpvorbispay, "unable to parse initial header");
gst_buffer_unmap (buf, &map);
return FALSE;
}
finish_failed:
{
GST_WARNING_OBJECT (rtpvorbispay, "unable to finish headers");
return FALSE;
}
}
static void
gst_rtp_vorbis_pay_reset_packet (GstRtpVorbisPay * rtpvorbispay, guint8 VDT)
{
guint payload_len;
GstRTPBuffer rtp = { NULL };
GST_LOG_OBJECT (rtpvorbispay, "reset packet");
rtpvorbispay->payload_pos = 4;
gst_rtp_buffer_map (rtpvorbispay->packet, GST_MAP_READ, &rtp);
payload_len = gst_rtp_buffer_get_payload_len (&rtp);
gst_rtp_buffer_unmap (&rtp);
rtpvorbispay->payload_left = payload_len - 4;
rtpvorbispay->payload_duration = 0;
rtpvorbispay->payload_F = 0;
rtpvorbispay->payload_VDT = VDT;
rtpvorbispay->payload_pkts = 0;
}
static void
gst_rtp_vorbis_pay_init_packet (GstRtpVorbisPay * rtpvorbispay, guint8 VDT,
GstClockTime timestamp)
{
GST_LOG_OBJECT (rtpvorbispay, "starting new packet, VDT: %d", VDT);
if (rtpvorbispay->packet)
gst_buffer_unref (rtpvorbispay->packet);
g_list_foreach (rtpvorbispay->packet_buffers, (GFunc) gst_mini_object_unref,
NULL);
g_list_free (rtpvorbispay->packet_buffers);
rtpvorbispay->packet_buffers = NULL;
/* new packet allocate max packet size */
rtpvorbispay->packet =
gst_rtp_buffer_new_allocate_len (GST_RTP_BASE_PAYLOAD_MTU
(rtpvorbispay), 0, 0);
gst_rtp_vorbis_pay_reset_packet (rtpvorbispay, VDT);
GST_BUFFER_PTS (rtpvorbispay->packet) = timestamp;
}
static GstFlowReturn
gst_rtp_vorbis_pay_flush_packet (GstRtpVorbisPay * rtpvorbispay)
{
GstFlowReturn ret;
guint8 *payload;
guint hlen;
GstRTPBuffer rtp = { NULL };
GList *l;
/* check for empty packet */
if (!rtpvorbispay->packet || rtpvorbispay->payload_pos <= 4)
return GST_FLOW_OK;
GST_LOG_OBJECT (rtpvorbispay, "flushing packet");
gst_rtp_buffer_map (rtpvorbispay->packet, GST_MAP_WRITE, &rtp);
/* fix header */
payload = gst_rtp_buffer_get_payload (&rtp);
/*
* 0 1 2 3
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | Ident | F |VDT|# pkts.|
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*
* F: Fragment type (0=none, 1=start, 2=cont, 3=end)
* VDT: Vorbis data type (0=vorbis, 1=config, 2=comment, 3=reserved)
* pkts: number of packets.
*/
payload[0] = (rtpvorbispay->payload_ident >> 16) & 0xff;
payload[1] = (rtpvorbispay->payload_ident >> 8) & 0xff;
payload[2] = (rtpvorbispay->payload_ident) & 0xff;
payload[3] = (rtpvorbispay->payload_F & 0x3) << 6 |
(rtpvorbispay->payload_VDT & 0x3) << 4 |
(rtpvorbispay->payload_pkts & 0xf);
gst_rtp_buffer_unmap (&rtp);
/* shrink the buffer size to the last written byte */
hlen = gst_rtp_buffer_calc_header_len (0);
gst_buffer_resize (rtpvorbispay->packet, 0, hlen + rtpvorbispay->payload_pos);
GST_BUFFER_DURATION (rtpvorbispay->packet) = rtpvorbispay->payload_duration;
for (l = g_list_last (rtpvorbispay->packet_buffers); l; l = l->prev) {
GstBuffer *buf = GST_BUFFER_CAST (l->data);
gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpvorbispay), rtpvorbispay->packet,
buf, g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
gst_buffer_unref (buf);
}
g_list_free (rtpvorbispay->packet_buffers);
rtpvorbispay->packet_buffers = NULL;
/* push, this gives away our ref to the packet, so clear it. */
ret =
gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpvorbispay),
rtpvorbispay->packet);
rtpvorbispay->packet = NULL;
return ret;
}
static gboolean
gst_rtp_vorbis_pay_finish_headers (GstRTPBasePayload * basepayload)
{
GstRtpVorbisPay *rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
GList *walk;
guint length, size, n_headers, configlen, extralen;
gchar *cstr, *configuration;
guint8 *data, *config;
guint32 ident;
gboolean res;
GST_DEBUG_OBJECT (rtpvorbispay, "finish headers");
if (!rtpvorbispay->headers)
goto no_headers;
/* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | Number of packed headers |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | Packed header |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | Packed header |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | .... |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*
* We only construct a config containing 1 packed header like this:
*
* 0 1 2 3
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | Ident | length ..
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* .. | n. of headers | length1 | length2 ..
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* .. | Identification Header ..
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* .................................................................
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* .. | Comment Header ..
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* .................................................................
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* .. Comment Header |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | Setup Header ..
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* .................................................................
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* .. Setup Header |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
/* we need 4 bytes for the number of headers (which is always 1), 3 bytes for
* the ident, 2 bytes for length, 1 byte for n. of headers. */
size = 4 + 3 + 2 + 1;
/* count the size of the headers first and update the hash */
length = 0;
n_headers = 0;
ident = fnv1_hash_32_new ();
extralen = 1;
for (walk = rtpvorbispay->headers; walk; walk = g_list_next (walk)) {
GstBuffer *buf = GST_BUFFER_CAST (walk->data);
GstMapInfo map;
guint bsize;
bsize = gst_buffer_get_size (buf);
length += bsize;
n_headers++;
/* count number of bytes needed for length fields, we don't need this for
* the last header. */
if (g_list_next (walk)) {
do {
size++;
extralen++;
bsize >>= 7;
} while (bsize);
}
/* update hash */
gst_buffer_map (buf, &map, GST_MAP_READ);
ident = fnv1_hash_32_update (ident, map.data, map.size);
gst_buffer_unmap (buf, &map);
}
/* packet length is header size + packet length */
configlen = size + length;
config = data = g_malloc (configlen);
/* number of packed headers, we only pack 1 header */
data[0] = 0;
data[1] = 0;
data[2] = 0;
data[3] = 1;
ident = fnv1_hash_32_to_24 (ident);
rtpvorbispay->payload_ident = ident;
GST_DEBUG_OBJECT (rtpvorbispay, "ident 0x%08x", ident);
/* take lower 3 bytes */
data[4] = (ident >> 16) & 0xff;
data[5] = (ident >> 8) & 0xff;
data[6] = ident & 0xff;
/* store length of all vorbis headers */
data[7] = ((length) >> 8) & 0xff;
data[8] = (length) & 0xff;
/* store number of headers minus one. */
data[9] = n_headers - 1;
data += 10;
/* store length for each header */
for (walk = rtpvorbispay->headers; walk; walk = g_list_next (walk)) {
GstBuffer *buf = GST_BUFFER_CAST (walk->data);
guint bsize, size, temp;
guint flag;
/* only need to store the length when it's not the last header */
if (!g_list_next (walk))
break;
bsize = gst_buffer_get_size (buf);
/* calc size */
size = 0;
do {
size++;
bsize >>= 7;
} while (bsize);
temp = size;
bsize = gst_buffer_get_size (buf);
/* write the size backwards */
flag = 0;
while (size) {
size--;
data[size] = (bsize & 0x7f) | flag;
bsize >>= 7;
flag = 0x80; /* Flag bit on all bytes of the length except the last */
}
data += temp;
}
/* copy header data */
for (walk = rtpvorbispay->headers; walk; walk = g_list_next (walk)) {
GstBuffer *buf = GST_BUFFER_CAST (walk->data);
gst_buffer_extract (buf, 0, data, gst_buffer_get_size (buf));
data += gst_buffer_get_size (buf);
}
rtpvorbispay->need_headers = FALSE;
/* serialize to base64 */
configuration = g_base64_encode (config, configlen);
/* store for later re-sending */
if (rtpvorbispay->config_data)
g_free (rtpvorbispay->config_data);
rtpvorbispay->config_size = configlen - 4 - 3 - 2;
rtpvorbispay->config_data = g_malloc (rtpvorbispay->config_size);
rtpvorbispay->config_extra_len = extralen;
memcpy (rtpvorbispay->config_data, config + 4 + 3 + 2,
rtpvorbispay->config_size);
g_free (config);
/* configure payloader settings */
cstr = g_strdup_printf ("%d", rtpvorbispay->channels);
gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "VORBIS",
rtpvorbispay->rate);
res =
gst_rtp_base_payload_set_outcaps (basepayload, "encoding-params",
G_TYPE_STRING, cstr, "configuration", G_TYPE_STRING, configuration, NULL);
g_free (cstr);
g_free (configuration);
return res;
/* ERRORS */
no_headers:
{
GST_DEBUG_OBJECT (rtpvorbispay, "finish headers");
return FALSE;
}
}
static gboolean
gst_rtp_vorbis_pay_parse_id (GstRTPBasePayload * basepayload, guint8 * data,
guint size)
{
GstRtpVorbisPay *rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
guint8 channels;
gint32 rate, version;
if (G_UNLIKELY (size < 16))
goto too_short;
if (G_UNLIKELY (memcmp (data, "\001vorbis", 7)))
goto invalid_start;
data += 7;
if (G_UNLIKELY ((version = GST_READ_UINT32_LE (data)) != 0))
goto invalid_version;
data += 4;
if (G_UNLIKELY ((channels = *data++) < 1))
goto invalid_channels;
if (G_UNLIKELY ((rate = GST_READ_UINT32_LE (data)) < 1))
goto invalid_rate;
/* all fine, store the values */
rtpvorbispay->channels = channels;
rtpvorbispay->rate = rate;
return TRUE;
/* ERRORS */
too_short:
{
GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
("Identification packet is too short, need at least 16, got %d", size),
(NULL));
return FALSE;
}
invalid_start:
{
GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
("Invalid header start in identification packet"), (NULL));
return FALSE;
}
invalid_version:
{
GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
("Invalid version, expected 0, got %d", version), (NULL));
return FALSE;
}
invalid_rate:
{
GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
("Invalid rate %d", rate), (NULL));
return FALSE;
}
invalid_channels:
{
GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
("Invalid channels %d", channels), (NULL));
return FALSE;
}
}
static GstFlowReturn
gst_rtp_vorbis_pay_payload_buffer (GstRtpVorbisPay * rtpvorbispay, guint8 VDT,
GstBuffer * buffer, guint8 * data, guint size, GstClockTime timestamp,
GstClockTime duration, guint not_in_length)
{
GstFlowReturn ret = GST_FLOW_OK;
guint newsize;
guint packet_len;
GstClockTime newduration;
gboolean flush;
guint plen;
guint8 *ppos, *payload;
gboolean fragmented;
GstRTPBuffer rtp = { NULL };
/* size increases with packet length and 2 bytes size eader. */
newduration = rtpvorbispay->payload_duration;
if (duration != GST_CLOCK_TIME_NONE)
newduration += duration;
newsize = rtpvorbispay->payload_pos + 2 + size;
packet_len = gst_rtp_buffer_calc_packet_len (newsize, 0, 0);
/* check buffer filled against length and max latency */
flush = gst_rtp_base_payload_is_filled (GST_RTP_BASE_PAYLOAD (rtpvorbispay),
packet_len, newduration);
/* we can store up to 15 vorbis packets in one RTP packet. */
flush |= (rtpvorbispay->payload_pkts == 15);
/* flush if we have a new VDT */
if (rtpvorbispay->packet)
flush |= (rtpvorbispay->payload_VDT != VDT);
if (flush)
ret = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay);
if (ret != GST_FLOW_OK)
goto done;
/* create new packet if we must */
if (!rtpvorbispay->packet) {
gst_rtp_vorbis_pay_init_packet (rtpvorbispay, VDT, timestamp);
}
gst_rtp_buffer_map (rtpvorbispay->packet, GST_MAP_WRITE, &rtp);
payload = gst_rtp_buffer_get_payload (&rtp);
ppos = payload + rtpvorbispay->payload_pos;
fragmented = FALSE;
/* put buffer in packet, it either fits completely or needs to be fragmented
* over multiple RTP packets. */
do {
plen = MIN (rtpvorbispay->payload_left - 2, size);
GST_LOG_OBJECT (rtpvorbispay, "append %u bytes", plen);
/* data is copied in the payload with a 2 byte length header */
ppos[0] = ((plen - not_in_length) >> 8) & 0xff;
ppos[1] = ((plen - not_in_length) & 0xff);
if (plen)
memcpy (&ppos[2], data, plen);
if (buffer) {
if (!rtpvorbispay->packet_buffers
|| rtpvorbispay->packet_buffers->data != (gpointer) buffer)
rtpvorbispay->packet_buffers =
g_list_prepend (rtpvorbispay->packet_buffers,
gst_buffer_ref (buffer));
} else {
GList *l;
for (l = rtpvorbispay->headers; l; l = l->next)
rtpvorbispay->packet_buffers =
g_list_prepend (rtpvorbispay->packet_buffers,
gst_buffer_ref (buffer));
}
/* only first (only) configuration cuts length field */
/* NOTE: spec (if any) is not clear on this ... */
not_in_length = 0;
size -= plen;
data += plen;
rtpvorbispay->payload_pos += plen + 2;
rtpvorbispay->payload_left -= plen + 2;
if (fragmented) {
if (size == 0)
/* last fragment, set F to 0x3. */
rtpvorbispay->payload_F = 0x3;
else
/* fragment continues, set F to 0x2. */
rtpvorbispay->payload_F = 0x2;
} else {
if (size > 0) {
/* fragmented packet starts, set F to 0x1, mark ourselves as
* fragmented. */
rtpvorbispay->payload_F = 0x1;
fragmented = TRUE;
}
}
if (fragmented) {
gst_rtp_buffer_unmap (&rtp);
/* fragmented packets are always flushed and have ptks of 0 */
rtpvorbispay->payload_pkts = 0;
ret = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay);
if (size > 0) {
/* start new packet and get pointers. VDT stays the same. */
gst_rtp_vorbis_pay_init_packet (rtpvorbispay,
rtpvorbispay->payload_VDT, timestamp);
gst_rtp_buffer_map (rtpvorbispay->packet, GST_MAP_WRITE, &rtp);
payload = gst_rtp_buffer_get_payload (&rtp);
ppos = payload + rtpvorbispay->payload_pos;
}
} else {
/* unfragmented packet, update stats for next packet, size == 0 and we
* exit the while loop */
rtpvorbispay->payload_pkts++;
if (duration != GST_CLOCK_TIME_NONE)
rtpvorbispay->payload_duration += duration;
}
} while (size && ret == GST_FLOW_OK);
if (rtp.buffer)
gst_rtp_buffer_unmap (&rtp);
done:
return ret;
}
static GstFlowReturn
gst_rtp_vorbis_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpVorbisPay *rtpvorbispay;
GstFlowReturn ret;
GstMapInfo map;
gsize size;
guint8 *data;
GstClockTime duration, timestamp;
guint8 VDT;
rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
gst_buffer_map (buffer, &map, GST_MAP_READ);
data = map.data;
size = map.size;
duration = GST_BUFFER_DURATION (buffer);
timestamp = GST_BUFFER_PTS (buffer);
GST_LOG_OBJECT (rtpvorbispay, "size %" G_GSIZE_FORMAT
", duration %" GST_TIME_FORMAT, size, GST_TIME_ARGS (duration));
if (G_UNLIKELY (size < 1))
goto wrong_size;
/* find packet type */
if (data[0] & 1) {
/* header */
if (data[0] == 1) {
/* identification, we need to parse this in order to get the clock rate. */
if (G_UNLIKELY (!gst_rtp_vorbis_pay_parse_id (basepayload, data, size)))
goto parse_id_failed;
VDT = 1;
} else if (data[0] == 3) {
/* comment */
VDT = 2;
} else if (data[0] == 5) {
/* setup */
VDT = 1;
} else
goto unknown_header;
} else
/* data */
VDT = 0;
/* we need to collect the headers and construct a config string from them */
if (VDT != 0) {
GST_DEBUG_OBJECT (rtpvorbispay, "collecting header");
/* append header to the list of headers */
gst_buffer_unmap (buffer, &map);
rtpvorbispay->headers = g_list_append (rtpvorbispay->headers, buffer);
ret = GST_FLOW_OK;
goto done;
} else if (rtpvorbispay->headers && rtpvorbispay->need_headers) {
if (!gst_rtp_vorbis_pay_finish_headers (basepayload))
goto header_error;
}
/* there is a config request, see if we need to insert it */
if (rtpvorbispay->config_interval > 0 && rtpvorbispay->config_data) {
gboolean send_config = FALSE;
if (rtpvorbispay->last_config != -1) {
guint64 diff;
GST_LOG_OBJECT (rtpvorbispay,
"now %" GST_TIME_FORMAT ", last config %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp), GST_TIME_ARGS (rtpvorbispay->last_config));
/* calculate diff between last config in milliseconds */
if (timestamp > rtpvorbispay->last_config) {
diff = timestamp - rtpvorbispay->last_config;
} else {
diff = 0;
}
GST_DEBUG_OBJECT (rtpvorbispay,
"interval since last config %" GST_TIME_FORMAT, GST_TIME_ARGS (diff));
/* bigger than interval, queue config */
/* FIXME should convert timestamps to running time */
if (GST_TIME_AS_SECONDS (diff) >= rtpvorbispay->config_interval) {
GST_DEBUG_OBJECT (rtpvorbispay, "time to send config");
send_config = TRUE;
}
} else {
/* no known previous config time, send now */
GST_DEBUG_OBJECT (rtpvorbispay, "no previous config time, send now");
send_config = TRUE;
}
if (send_config) {
/* we need to send config now first */
/* different TDT type forces flush */
gst_rtp_vorbis_pay_payload_buffer (rtpvorbispay, 1,
NULL, rtpvorbispay->config_data, rtpvorbispay->config_size,
timestamp, GST_CLOCK_TIME_NONE, rtpvorbispay->config_extra_len);
if (timestamp != -1) {
rtpvorbispay->last_config = timestamp;
}
}
}
ret =
gst_rtp_vorbis_pay_payload_buffer (rtpvorbispay, VDT, buffer, data, size,
timestamp, duration, 0);
gst_buffer_unmap (buffer, &map);
gst_buffer_unref (buffer);
done:
return ret;
/* ERRORS */
wrong_size:
{
GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE,
("Invalid packet size (1 < %" G_GSIZE_FORMAT ")", size), (NULL));
gst_buffer_unmap (buffer, &map);
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
parse_id_failed:
{
gst_buffer_unmap (buffer, &map);
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
unknown_header:
{
GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE,
(NULL), ("Ignoring unknown header received"));
gst_buffer_unmap (buffer, &map);
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
header_error:
{
GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE,
(NULL), ("Error initializing header config"));
gst_buffer_unmap (buffer, &map);
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
}
static gboolean
gst_rtp_vorbis_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
{
GstRtpVorbisPay *rtpvorbispay = GST_RTP_VORBIS_PAY (payload);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
gst_rtp_vorbis_pay_clear_packet (rtpvorbispay);
break;
default:
break;
}
/* false to let parent handle event as well */
return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
}
static GstStateChangeReturn
gst_rtp_vorbis_pay_change_state (GstElement * element,
GstStateChange transition)
{
GstRtpVorbisPay *rtpvorbispay;
GstStateChangeReturn ret;
rtpvorbispay = GST_RTP_VORBIS_PAY (element);
switch (transition) {
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtp_vorbis_pay_cleanup (rtpvorbispay);
break;
default:
break;
}
return ret;
}
static void
gst_rtp_vorbis_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpVorbisPay *rtpvorbispay;
rtpvorbispay = GST_RTP_VORBIS_PAY (object);
switch (prop_id) {
case PROP_CONFIG_INTERVAL:
rtpvorbispay->config_interval = g_value_get_uint (value);
break;
default:
break;
}
}
static void
gst_rtp_vorbis_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpVorbisPay *rtpvorbispay;
rtpvorbispay = GST_RTP_VORBIS_PAY (object);
switch (prop_id) {
case PROP_CONFIG_INTERVAL:
g_value_set_uint (value, rtpvorbispay->config_interval);
break;
default:
break;
}
}
gboolean
gst_rtp_vorbis_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpvorbispay",
GST_RANK_SECONDARY, GST_TYPE_RTP_VORBIS_PAY);
}
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