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gstreamer/gst/rtp/gstrtpvorbispay.c
Wim Taymans 20f18abf72 gst/rtp/: Fix encoding-name case. 
Original commit message from CVS:
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_setcaps):
* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_finish_headers):
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers):
Fix encoding-name case.
2007-03-05 17:08:32 +00:00

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/* GStreamer
* Copyright (C) <2006> Wim Taymans <wim@fluendo.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpvorbispay.h"
GST_DEBUG_CATEGORY_STATIC (rtpvorbispay_debug);
#define GST_CAT_DEFAULT (rtpvorbispay_debug)
/* references:
* http://svn.xiph.org/trunk/vorbis/doc/draft-ietf-avt-rtp-vorbis-01.txt
*/
/* elementfactory information */
static const GstElementDetails gst_rtp_vorbispay_details =
GST_ELEMENT_DETAILS ("RTP packet depayloader",
"Codec/Payloader/Network",
"Payload-encode Vorbis audio into RTP packets (draft-01 RFC XXXX)",
"Wim Taymans <wim@fluendo.com>");
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>"
* "delivery-method = (string) { inline, in_band, out_band/<specific_name> } "
* "configuration = (string) ANY"
*/
/* All optional parameters
*
* "configuration-uri ="
*/
)
);
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")
);
GST_BOILERPLATE (GstRtpVorbisPay, gst_rtp_vorbis_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
static gboolean gst_rtp_vorbis_pay_setcaps (GstBaseRTPPayload * basepayload,
GstCaps * caps);
static GstFlowReturn gst_rtp_vorbis_pay_handle_buffer (GstBaseRTPPayload * pad,
GstBuffer * buffer);
static void
gst_rtp_vorbis_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_vorbis_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_vorbis_pay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_vorbispay_details);
}
static void
gst_rtp_vorbis_pay_class_init (GstRtpVorbisPayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gstbasertppayload_class->set_caps = gst_rtp_vorbis_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_vorbis_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtpvorbispay_debug, "rtpvorbispay", 0,
"Vorbis RTP Payloader");
}
static void
gst_rtp_vorbis_pay_init (GstRtpVorbisPay * rtpvorbispay,
GstRtpVorbisPayClass * klass)
{
}
static gboolean
gst_rtp_vorbis_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
{
GstRtpVorbisPay *rtpvorbispay;
rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
rtpvorbispay->need_headers = TRUE;
return TRUE;
}
static void
gst_rtp_vorbis_pay_reset_packet (GstRtpVorbisPay * rtpvorbispay, guint8 VDT)
{
guint payload_len;
GST_DEBUG_OBJECT (rtpvorbispay, "reset packet");
rtpvorbispay->payload_pos = 4;
payload_len = gst_rtp_buffer_get_payload_len (rtpvorbispay->packet);
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_DEBUG_OBJECT (rtpvorbispay, "starting new packet, VDT: %d", VDT);
if (rtpvorbispay->packet)
gst_buffer_unref (rtpvorbispay->packet);
/* new packet allocate max packet size */
rtpvorbispay->packet =
gst_rtp_buffer_new_allocate_len (GST_BASE_RTP_PAYLOAD_MTU
(rtpvorbispay), 0, 0);
gst_rtp_vorbis_pay_reset_packet (rtpvorbispay, VDT);
GST_BUFFER_TIMESTAMP (rtpvorbispay->packet) = timestamp;
}
static GstFlowReturn
gst_rtp_vorbis_pay_flush_packet (GstRtpVorbisPay * rtpvorbispay)
{
GstFlowReturn ret;
guint8 *payload;
guint hlen;
/* check for empty packet */
if (!rtpvorbispay->packet || rtpvorbispay->payload_pos <= 4)
return GST_FLOW_OK;
GST_DEBUG_OBJECT (rtpvorbispay, "flushing packet");
/* fix header */
payload = gst_rtp_buffer_get_payload (rtpvorbispay->packet);
/*
* 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);
/* shrink the buffer size to the last written byte */
hlen = gst_rtp_buffer_calc_header_len (0);
GST_BUFFER_SIZE (rtpvorbispay->packet) = hlen + rtpvorbispay->payload_pos;
/* push, this gives away our ref to the packet, so clear it. */
ret =
gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpvorbispay),
rtpvorbispay->packet);
rtpvorbispay->packet = NULL;
return ret;
}
static gboolean
gst_rtp_vorbis_pay_finish_headers (GstBaseRTPPayload * basepayload)
{
GstRtpVorbisPay *rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
GList *walk;
guint length;
gchar *cstr, *configuration;
guint8 *data;
guint32 ident;
GValue v = { 0 };
GstBuffer *config;
GST_DEBUG_OBJECT (rtpvorbispay, "finish headers");
if (!rtpvorbispay->headers)
goto no_headers;
/* we need exactly 2 header packets */
if (g_list_length (rtpvorbispay->headers) != 2)
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 | Identification Header ..
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* .. Identification Header |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | Setup Header ..
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* .. Setup Header |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*
*/
/* count the size of the headers first */
length = 0;
for (walk = rtpvorbispay->headers; walk; walk = g_list_next (walk)) {
GstBuffer *buf = GST_BUFFER_CAST (walk->data);
length += GST_BUFFER_SIZE (buf);
}
/* total config length is 4 bytes header number + size of the
* headers + 2 bytes length + 3 bytes for the ident */
config = gst_buffer_new_and_alloc (length + 4 + 2 + 3);
data = GST_BUFFER_DATA (config);
/* number of packed headers, we only pack 1 header */
data[0] = 0;
data[1] = 0;
data[2] = 0;
data[3] = 1;
/* we generate a random ident for this configuration */
ident = rtpvorbispay->payload_ident = g_random_int ();
/* take lower 3 bytes */
data[4] = (ident >> 16) & 0xff;
data[5] = (ident >> 8) & 0xff;
data[6] = ident & 0xff;
/* store length minus the length of the fixed vorbis header */
data[7] = ((length - 30) >> 8) & 0xff;
data[8] = (length - 30) & 0xff;
/* copy header data */
data += 9;
for (walk = rtpvorbispay->headers; walk; walk = g_list_next (walk)) {
GstBuffer *buf = GST_BUFFER_CAST (walk->data);
memcpy (data, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
data += GST_BUFFER_SIZE (buf);
}
/* serialize buffer to base16 */
g_value_init (&v, GST_TYPE_BUFFER);
gst_value_take_buffer (&v, config);
configuration = gst_value_serialize (&v);
/* configure payloader settings */
cstr = g_strdup_printf ("%d", rtpvorbispay->channels);
gst_basertppayload_set_options (basepayload, "audio", TRUE, "VORBIS",
rtpvorbispay->rate);
gst_basertppayload_set_outcaps (basepayload, "encoding-params", G_TYPE_STRING,
cstr, "configuration", G_TYPE_STRING, configuration,
"delivery-method", G_TYPE_STRING, "inline",
/* don't set the other defaults
*/
NULL);
g_free (cstr);
g_free (configuration);
g_value_unset (&v);
return TRUE;
/* ERRORS */
no_headers:
{
GST_DEBUG_OBJECT (rtpvorbispay, "finish headers");
return FALSE;
}
}
static gboolean
gst_rtp_vorbis_pay_parse_id (GstBaseRTPPayload * 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_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpVorbisPay *rtpvorbispay;
GstFlowReturn ret;
guint size, newsize;
guint8 *data;
guint packet_len;
GstClockTime duration, newduration, timestamp;
gboolean flush;
guint8 VDT;
guint plen;
guint8 *ppos, *payload;
gboolean fragmented;
rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
duration = GST_BUFFER_DURATION (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
GST_DEBUG_OBJECT (rtpvorbispay, "size %u, duration %" GST_TIME_FORMAT,
size, GST_TIME_ARGS (duration));
if (G_UNLIKELY (size < 1 || size > 0xffff))
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;
if (rtpvorbispay->need_headers) {
/* we need to collect the headers and construct a config string from them */
if (VDT == 2) {
GST_DEBUG_OBJECT (rtpvorbispay,
"discard comment packet while collecting headers");
ret = GST_FLOW_OK;
goto done;
} else if (VDT != 0) {
GST_DEBUG_OBJECT (rtpvorbispay, "collecting header");
/* append header to the list of headers */
rtpvorbispay->headers = g_list_append (rtpvorbispay->headers, buffer);
ret = GST_FLOW_OK;
goto done;
} else {
if (!gst_rtp_vorbis_pay_finish_headers (basepayload))
goto header_error;
rtpvorbispay->need_headers = FALSE;
}
}
/* 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_basertppayload_is_filled (basepayload, 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);
/* create new packet if we must */
if (!rtpvorbispay->packet) {
gst_rtp_vorbis_pay_init_packet (rtpvorbispay, VDT, timestamp);
}
payload = gst_rtp_buffer_get_payload (rtpvorbispay->packet);
ppos = payload + rtpvorbispay->payload_pos;
fragmented = FALSE;
ret = GST_FLOW_OK;
/* put buffer in packet, it either fits completely or needs to be fragmented
* over multiple RTP packets. */
while (size) {
plen = MIN (rtpvorbispay->payload_left - 2, size);
GST_DEBUG_OBJECT (rtpvorbispay, "append %u bytes", plen);
/* data is copied in the payload with a 2 byte length header */
ppos[0] = (plen >> 8) & 0xff;
ppos[1] = (plen & 0xff);
memcpy (&ppos[2], data, plen);
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) {
/* 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);
payload = gst_rtp_buffer_get_payload (rtpvorbispay->packet);
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;
}
}
done:
return ret;
/* ERRORS */
wrong_size:
{
GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE,
("Invalid packet size (1 < %d <= 0xffff)", size), (NULL));
return GST_FLOW_OK;
}
parse_id_failed:
{
return GST_FLOW_ERROR;
}
unknown_header:
{
GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE,
(NULL), ("Ignoring unknown header received"));
return GST_FLOW_OK;
}
header_error:
{
GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE,
(NULL), ("Error initializing header config"));
return GST_FLOW_OK;
}
}
gboolean
gst_rtp_vorbis_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpvorbispay",
GST_RANK_NONE, GST_TYPE_RTP_VORBIS_PAY);
}
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