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gstreamer/gst/rtp/gstrtpmp4gpay.c
Thomas Vander Stichele c3fac62611 fix descriptions and license blocks cut and paste anyone ? 
Original commit message from CVS:

fix descriptions and license blocks
cut and paste anyone ?
2006-05-22 13:51:30 +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 "gstrtpmp4gpay.h"
GST_DEBUG_CATEGORY (rtpmp4gpay_debug);
#define GST_CAT_DEFAULT (rtpmp4gpay_debug)
/* elementfactory information */
static const GstElementDetails gst_rtp_mp4gpay_details =
GST_ELEMENT_DETAILS ("RTP packet parser",
"Codec/Payloader/Network",
"Payload MPEG4 elementary streams as RTP packets (RFC 3640)",
"Wim Taymans <wim@fluendo.com>");
static GstStaticPadTemplate gst_rtp_mp4g_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/mpeg,"
"mpegversion=(int) 4,"
"systemstream=(boolean)false;" "audio/mpeg," "mpegversion=(int) 4")
);
static GstStaticPadTemplate gst_rtp_mp4g_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) [ 96, 127 ], "
"clock-rate = (int) [1, MAX ], "
"encoding-name = (string) \"mpeg4-generic\", "
/* required string params */
"streamtype = (string) { \"4\", \"5\" }, " /* 4 = video, 5 = audio */
"profile-level-id = (int) [1,MAX], "
/* "config = (string) [1,MAX]" */
"mode = (string) { \"generic\", \"CELP-cbr\", \"CELP-vbr\", \"AAC-lbr\", \"AAC-hbr\" }, "
/* Optional general parameters */
"objecttype = (int) [1,MAX], " "constantsize = (int) [1,MAX], " /* constant size of each AU */
"constantduration = (int) [1,MAX], " /* constant duration of each AU */
"maxdisplacement = (int) [1,MAX], "
"de-interleavebuffersize: = (int) [1,MAX], "
/* Optional configuration parameters */
"sizelength = (int) [1, 16], " /* max 16 bits, should be enough... */
"indexlength = (int) [1, 8], "
"indexdeltalength = (int) [1, 8], "
"ctsdeltalength = (int) [1, 64], "
"dtsdeltalength = (int) [1, 64], "
"randomaccessindication = (int) {0, 1}, "
"streamstateindication = (int) [0, 64], "
"auxiliarydatasizelength = (int) [0, 64]")
);
enum
{
PROP_0,
};
static void gst_rtp_mp4g_pay_class_init (GstRtpMP4GPayClass * klass);
static void gst_rtp_mp4g_pay_base_init (GstRtpMP4GPayClass * klass);
static void gst_rtp_mp4g_pay_init (GstRtpMP4GPay * rtpmp4gpay);
static void gst_rtp_mp4g_pay_finalize (GObject * object);
static void gst_rtp_mp4g_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_mp4g_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
static GstBaseRTPPayloadClass *parent_class = NULL;
static GType
gst_rtp_mp4g_pay_get_type (void)
{
static GType rtpmp4gpay_type = 0;
if (!rtpmp4gpay_type) {
static const GTypeInfo rtpmp4gpay_info = {
sizeof (GstRtpMP4GPayClass),
(GBaseInitFunc) gst_rtp_mp4g_pay_base_init,
NULL,
(GClassInitFunc) gst_rtp_mp4g_pay_class_init,
NULL,
NULL,
sizeof (GstRtpMP4GPay),
0,
(GInstanceInitFunc) gst_rtp_mp4g_pay_init,
};
rtpmp4gpay_type =
g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpMP4GPay",
&rtpmp4gpay_info, 0);
}
return rtpmp4gpay_type;
}
static void
gst_rtp_mp4g_pay_base_init (GstRtpMP4GPayClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_mp4g_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_mp4g_pay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_mp4gpay_details);
}
static void
gst_rtp_mp4g_pay_class_init (GstRtpMP4GPayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->set_property = gst_rtp_mp4g_pay_set_property;
gobject_class->get_property = gst_rtp_mp4g_pay_get_property;
gobject_class->finalize = gst_rtp_mp4g_pay_finalize;
gstbasertppayload_class->set_caps = gst_rtp_mp4g_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_mp4g_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtpmp4gpay_debug, "rtpmp4gpay", 0,
"MP4-generic RTP Payloader");
}
static void
gst_rtp_mp4g_pay_init (GstRtpMP4GPay * rtpmp4gpay)
{
rtpmp4gpay->adapter = gst_adapter_new ();
rtpmp4gpay->rate = 90000;
rtpmp4gpay->profile = 1;
rtpmp4gpay->mode = "";
}
static void
gst_rtp_mp4g_pay_finalize (GObject * object)
{
GstRtpMP4GPay *rtpmp4gpay;
rtpmp4gpay = GST_RTP_MP4G_PAY (object);
g_object_unref (rtpmp4gpay->adapter);
rtpmp4gpay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static unsigned sampling_table[16] = {
96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
16000, 12000, 11025, 8000, 7350, 0, 0, 0
};
static gboolean
gst_rtp_mp4g_pay_parse_audio_config (GstRtpMP4GPay * rtpmp4gpay,
GstBuffer * buffer)
{
guint8 *data;
guint size;
guint8 objectType;
guint8 samplingIdx;
guint8 channelCfg;
data = GST_BUFFER_DATA (buffer);
size = GST_BUFFER_SIZE (buffer);
if (size < 2)
goto too_short;
/* only AAC LC for now */
objectType = (data[0] & 0xf8) >> 3;
if (objectType != 2)
goto unsupported_type;
samplingIdx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7);
/* only fixed values for now */
if (samplingIdx > 12 && samplingIdx != 15)
goto wrong_freq;
channelCfg = ((data[1] & 0x78) >> 3);
if (channelCfg > 7)
goto wrong_channels;
/* rtp rate depends on sampling rate of the audio */
if (samplingIdx == 15) {
if (size < 5)
goto too_short;
/* index of 15 means we get the rate in the next 24 bits */
rtpmp4gpay->rate = ((data[1] & 0x7f) << 17) |
((data[2]) << 9) | ((data[3]) << 1) | ((data[4] & 0x80) >> 7);
} else {
/* else use the rate from the table */
rtpmp4gpay->rate = sampling_table[samplingIdx];
}
/* extra rtp params contain the number of channels */
rtpmp4gpay->params = channelCfg;
/* audio stream type */
rtpmp4gpay->streamtype = 5;
/* mode */
rtpmp4gpay->mode = "ACC-hbr";
/* profile (should be 1) */
rtpmp4gpay->profile = objectType - 1;
GST_DEBUG_OBJECT (rtpmp4gpay,
"objectType: %d, samplingIdx: %d (%d), channelCfg: %d", objectType,
samplingIdx, rtpmp4gpay->rate, channelCfg);
return TRUE;
/* ERROR */
too_short:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
(NULL), ("config string too short"));
return FALSE;
}
unsupported_type:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
(NULL), ("unsupported object type %d", objectType));
return FALSE;
}
wrong_freq:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
(NULL), ("unsupported frequency index %d", samplingIdx));
return FALSE;
}
wrong_channels:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
(NULL), ("unsupported number of channels %d", channelCfg));
return FALSE;
}
}
#define VOS_STARTCODE 0x000001B0
static gboolean
gst_rtp_mp4g_pay_parse_video_config (GstRtpMP4GPay * rtpmp4gpay,
GstBuffer * buffer)
{
guint8 *data;
guint size;
guint32 code;
data = GST_BUFFER_DATA (buffer);
size = GST_BUFFER_SIZE (buffer);
if (size < 5)
goto too_short;
code = GST_READ_UINT32_BE (data);
if (code == VOS_STARTCODE) {
/* get profile */
rtpmp4gpay->profile = data[4];
} else {
GST_ELEMENT_WARNING (rtpmp4gpay, STREAM, FORMAT,
(NULL), ("profile not found in config string"));
rtpmp4gpay->profile = 1;
}
/* fixed rate */
rtpmp4gpay->rate = 90000;
/* video stream type */
rtpmp4gpay->streamtype = 4;
/* no params for video */
rtpmp4gpay->params = 0;
/* mode */
rtpmp4gpay->mode = "generic";
GST_LOG_OBJECT (rtpmp4gpay, "profile %d", rtpmp4gpay->profile);
return TRUE;
/* ERROR */
too_short:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
(NULL), ("config string too short"));
return FALSE;
}
}
static void
gst_rtp_mp4g_pay_new_caps (GstRtpMP4GPay * rtpmp4gpay)
{
gchar *config;
GValue v = { 0 };
#define MP4GCAPS \
"streamtype", G_TYPE_INT, rtpmp4gpay->streamtype, \
"profile-level-id", G_TYPE_INT, rtpmp4gpay->profile, \
"mode", G_TYPE_STRING, rtpmp4gpay->mode, \
"config", G_TYPE_STRING, config, \
"sizelength", G_TYPE_INT, 13, \
"indexlength", G_TYPE_INT, 3, \
"indexdeltalength", G_TYPE_INT, 3, \
NULL
g_value_init (&v, GST_TYPE_BUFFER);
gst_value_set_buffer (&v, rtpmp4gpay->config);
config = gst_value_serialize (&v);
/* hmm, silly */
if (rtpmp4gpay->params) {
gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
"encoding-params", G_TYPE_INT, rtpmp4gpay->params, MP4GCAPS);
} else {
gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
MP4GCAPS);
}
g_value_unset (&v);
g_free (config);
#undef MP4GCAPS
}
static gboolean
gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
GstRtpMP4GPay *rtpmp4gpay;
GstStructure *structure;
const GValue *codec_data;
gboolean res = TRUE;
rtpmp4gpay = GST_RTP_MP4G_PAY (payload);
structure = gst_caps_get_structure (caps, 0);
codec_data = gst_structure_get_value (structure, "codec_data");
if (codec_data) {
GST_LOG_OBJECT (rtpmp4gpay, "got codec_data");
if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) {
GstBuffer *buffer;
const gchar *name;
buffer = gst_value_get_buffer (codec_data);
GST_LOG_OBJECT (rtpmp4gpay, "configuring codec_data");
name = gst_structure_get_name (structure);
/* parse buffer */
if (!strcmp (name, "audio/mpeg")) {
res = gst_rtp_mp4g_pay_parse_audio_config (rtpmp4gpay, buffer);
} else if (!strcmp (name, "video/mpeg")) {
res = gst_rtp_mp4g_pay_parse_video_config (rtpmp4gpay, buffer);
} else {
res = FALSE;
}
if (!res)
goto config_failed;
/* now we can configure the buffer */
if (rtpmp4gpay->config)
gst_buffer_unref (rtpmp4gpay->config);
rtpmp4gpay->config = gst_buffer_copy (buffer);
}
}
gst_basertppayload_set_options (payload, "video", TRUE, "mpeg4-generic",
rtpmp4gpay->rate);
gst_rtp_mp4g_pay_new_caps (rtpmp4gpay);
return res;
/* ERRORS */
config_failed:
{
GST_DEBUG_OBJECT (rtpmp4gpay, "failed to parse config");
return FALSE;
}
}
static GstFlowReturn
gst_rtp_mp4g_pay_flush (GstRtpMP4GPay * rtpmp4gpay)
{
guint avail, total;
GstBuffer *outbuf;
GstFlowReturn ret;
gboolean fragmented;
guint mtu;
fragmented = FALSE;
/* the data available in the adapter is either smaller
* than the MTU or bigger. In the case it is smaller, the complete
* adapter contents can be put in one packet. In the case the
* adapter has more than one MTU, we need to fragment the MPEG data
* over multiple packets. */
total = avail = gst_adapter_available (rtpmp4gpay->adapter);
ret = GST_FLOW_OK;
mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpmp4gpay);
while (avail > 0) {
guint towrite;
guint8 *payload;
guint8 *data;
guint payload_len;
guint packet_len;
/* this will be the total lenght of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0);
/* fill one MTU or all available bytes, we need 4 spare bytes for
* the AU header. */
towrite = MIN (packet_len, mtu - 4);
/* this is the payload length */
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
GST_DEBUG_OBJECT (rtpmp4gpay,
"avail %d, towrite %d, packet_len %d, payload_len %d", avail, towrite,
packet_len, payload_len);
/* create buffer to hold the payload, also make room for the 4 header bytes. */
outbuf = gst_rtp_buffer_new_allocate (payload_len + 4, 0, 0);
/* copy payload */
payload = gst_rtp_buffer_get_payload (outbuf);
/* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
* |AU-headers-length|AU-header|AU-header| |AU-header|padding|
* | | (1) | (2) | | (n) | bits |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
*/
/* AU-headers-length, we only have 1 AU-header */
payload[0] = 0x00;
payload[1] = 0x10; /* we use 16 bits for the header */
/* +---------------------------------------+
* | AU-size |
* +---------------------------------------+
* | AU-Index / AU-Index-delta |
* +---------------------------------------+
* | CTS-flag |
* +---------------------------------------+
* | CTS-delta |
* +---------------------------------------+
* | DTS-flag |
* +---------------------------------------+
* | DTS-delta |
* +---------------------------------------+
* | RAP-flag |
* +---------------------------------------+
* | Stream-state |
* +---------------------------------------+
*/
/* The AU-header, no CTS, DTS, RAP, Stream-state
*
* AU-size is always the total size of the AU, not the fragmented size
*/
payload[2] = (total & 0x1fe0) >> 5;
payload[3] = (total & 0x1f) << 3; /* we use 13 bits for the size, 3 bits index */
data = (guint8 *) gst_adapter_peek (rtpmp4gpay->adapter, payload_len);
memcpy (&payload[4], data, payload_len);
gst_adapter_flush (rtpmp4gpay->adapter, payload_len);
/* marker only if the packet is complete */
gst_rtp_buffer_set_marker (outbuf, avail > payload_len);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4gpay->first_ts;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4gpay), outbuf);
avail -= payload_len;
fragmented = TRUE;
}
return ret;
}
/* we expect buffers as exactly one complete AU
*/
static GstFlowReturn
gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpMP4GPay *rtpmp4gpay;
GstFlowReturn ret;
ret = GST_FLOW_OK;
rtpmp4gpay = GST_RTP_MP4G_PAY (basepayload);
rtpmp4gpay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
/* we always encode and flush a full AU */
gst_adapter_push (rtpmp4gpay->adapter, buffer);
ret = gst_rtp_mp4g_pay_flush (rtpmp4gpay);
return ret;
}
static void
gst_rtp_mp4g_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpMP4GPay *rtpmp4gpay;
rtpmp4gpay = GST_RTP_MP4G_PAY (object);
switch (prop_id) {
default:
break;
}
}
static void
gst_rtp_mp4g_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpMP4GPay *rtpmp4gpay;
rtpmp4gpay = GST_RTP_MP4G_PAY (object);
switch (prop_id) {
default:
break;
}
}
gboolean
gst_rtp_mp4g_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpmp4gpay",
GST_RANK_NONE, GST_TYPE_RTP_MP4G_PAY);
}
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