mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-25 03:01:03 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/rtp/gstrtpmp4gpay.c
Mark Nauwelaerts eb82a50bd1 rtp: port remaining to 0.11
2011-07-10 21:50:19 +02:00

636 lines
18 KiB
C
Raw Blame History

/* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/base/gstbitreader.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpmp4gpay.h"
GST_DEBUG_CATEGORY_STATIC (rtpmp4gpay_debug);
#define GST_CAT_DEFAULT (rtpmp4gpay_debug)
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, " "stream-format=(string) raw")
);
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\", \"audio\", \"application\" }, "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) [1, MAX ], "
"encoding-name = (string) \"MPEG4-GENERIC\", "
/* required string params */
"streamtype = (string) { \"4\", \"5\" }, " /* 4 = video, 5 = audio */
/* "profile-level-id = (string) [1,MAX], " */
/* "config = (string) [1,MAX]" */
"mode = (string) { \"generic\", \"CELP-cbr\", \"CELP-vbr\", \"AAC-lbr\", \"AAC-hbr\" } "
/* Optional general parameters */
/* "objecttype = (string) [1,MAX], " */
/* "constantsize = (string) [1,MAX], " *//* constant size of each AU */
/* "constantduration = (string) [1,MAX], " *//* constant duration of each AU */
/* "maxdisplacement = (string) [1,MAX], " */
/* "de-interleavebuffersize = (string) [1,MAX], " */
/* Optional configuration parameters */
/* "sizelength = (string) [1, 16], " *//* max 16 bits, should be enough... */
/* "indexlength = (string) [1, 8], " */
/* "indexdeltalength = (string) [1, 8], " */
/* "ctsdeltalength = (string) [1, 64], " */
/* "dtsdeltalength = (string) [1, 64], " */
/* "randomaccessindication = (string) {0, 1}, " */
/* "streamstateindication = (string) [0, 64], " */
/* "auxiliarydatasizelength = (string) [0, 64]" */ )
);
static void gst_rtp_mp4g_pay_finalize (GObject * object);
static GstStateChangeReturn gst_rtp_mp4g_pay_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
static gboolean gst_rtp_mp4g_pay_handle_event (GstBaseRTPPayload * payload,
GstEvent * event);
#define gst_rtp_mp4g_pay_parent_class parent_class
G_DEFINE_TYPE (GstRtpMP4GPay, gst_rtp_mp4g_pay, GST_TYPE_BASE_RTP_PAYLOAD)
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;
gobject_class->finalize = gst_rtp_mp4g_pay_finalize;
gstelement_class->change_state = gst_rtp_mp4g_pay_change_state;
gstbasertppayload_class->set_caps = gst_rtp_mp4g_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_mp4g_pay_handle_buffer;
gstbasertppayload_class->handle_event = gst_rtp_mp4g_pay_handle_event;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_mp4g_pay_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_mp4g_pay_sink_template));
gst_element_class_set_details_simple (gstelement_class,
"RTP MPEG4 ES payloader",
"Codec/Payloader/Network/RTP",
"Payload MPEG4 elementary streams as RTP packets (RFC 3640)",
"Wim Taymans <wim.taymans@gmail.com>");
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 ();
}
static void
gst_rtp_mp4g_pay_reset (GstRtpMP4GPay * rtpmp4gpay)
{
GST_DEBUG_OBJECT (rtpmp4gpay, "reset");
gst_adapter_clear (rtpmp4gpay->adapter);
rtpmp4gpay->offset = 0;
}
static void
gst_rtp_mp4g_pay_cleanup (GstRtpMP4GPay * rtpmp4gpay)
{
gst_rtp_mp4g_pay_reset (rtpmp4gpay);
g_free (rtpmp4gpay->params);
rtpmp4gpay->params = NULL;
if (rtpmp4gpay->config)
gst_buffer_unref (rtpmp4gpay->config);
rtpmp4gpay->config = NULL;
g_free (rtpmp4gpay->profile);
rtpmp4gpay->profile = NULL;
rtpmp4gpay->streamtype = NULL;
rtpmp4gpay->mode = NULL;
rtpmp4gpay->frame_len = 0;
}
static void
gst_rtp_mp4g_pay_finalize (GObject * object)
{
GstRtpMP4GPay *rtpmp4gpay;
rtpmp4gpay = GST_RTP_MP4G_PAY (object);
gst_rtp_mp4g_pay_cleanup (rtpmp4gpay);
g_object_unref (rtpmp4gpay->adapter);
rtpmp4gpay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static const unsigned int 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;
gsize size;
guint8 objectType = 0;
guint8 samplingIdx = 0;
guint8 channelCfg = 0;
GstBitReader br;
data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
gst_bit_reader_init (&br, data, size);
/* any object type is fine, we need to copy it to the profile-level-id field. */
if (!gst_bit_reader_get_bits_uint8 (&br, &objectType, 5))
goto too_short;
if (objectType == 0)
goto invalid_object;
if (!gst_bit_reader_get_bits_uint8 (&br, &samplingIdx, 4))
goto too_short;
/* only fixed values for now */
if (samplingIdx > 12 && samplingIdx != 15)
goto wrong_freq;
if (!gst_bit_reader_get_bits_uint8 (&br, &channelCfg, 4))
goto too_short;
if (channelCfg > 7)
goto wrong_channels;
/* rtp rate depends on sampling rate of the audio */
if (samplingIdx == 15) {
guint32 rate = 0;
/* index of 15 means we get the rate in the next 24 bits */
if (!gst_bit_reader_get_bits_uint32 (&br, &rate, 24))
goto too_short;
rtpmp4gpay->rate = rate;
} else {
/* else use the rate from the table */
rtpmp4gpay->rate = sampling_table[samplingIdx];
}
rtpmp4gpay->frame_len = 1024;
switch (objectType) {
case 1:
case 2:
case 3:
case 4:
case 6:
case 7:
{
guint8 frameLenFlag = 0;
if (gst_bit_reader_get_bits_uint8 (&br, &frameLenFlag, 1))
if (frameLenFlag)
rtpmp4gpay->frame_len = 960;
break;
}
default:
break;
}
/* extra rtp params contain the number of channels */
g_free (rtpmp4gpay->params);
rtpmp4gpay->params = g_strdup_printf ("%d", channelCfg);
/* audio stream type */
rtpmp4gpay->streamtype = "5";
/* mode only high bitrate for now */
rtpmp4gpay->mode = "AAC-hbr";
/* profile */
g_free (rtpmp4gpay->profile);
rtpmp4gpay->profile = g_strdup_printf ("%d", objectType);
GST_DEBUG_OBJECT (rtpmp4gpay,
"objectType: %d, samplingIdx: %d (%d), channelCfg: %d, frame_len %d",
objectType, samplingIdx, rtpmp4gpay->rate, channelCfg,
rtpmp4gpay->frame_len);
gst_buffer_unmap (buffer, data, -1);
return TRUE;
/* ERROR */
too_short:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
(NULL), ("config string too short"));
gst_buffer_unmap (buffer, data, -1);
return FALSE;
}
invalid_object:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
(NULL), ("invalid object type"));
gst_buffer_unmap (buffer, data, -1);
return FALSE;
}
wrong_freq:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
(NULL), ("unsupported frequency index %d", samplingIdx));
gst_buffer_unmap (buffer, data, -1);
return FALSE;
}
wrong_channels:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
(NULL), ("unsupported number of channels %d, must < 8", channelCfg));
gst_buffer_unmap (buffer, data, -1);
return FALSE;
}
}
#define VOS_STARTCODE 0x000001B0
static gboolean
gst_rtp_mp4g_pay_parse_video_config (GstRtpMP4GPay * rtpmp4gpay,
GstBuffer * buffer)
{
guint8 *data;
gsize size;
guint32 code;
data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
if (size < 5)
goto too_short;
code = GST_READ_UINT32_BE (data);
g_free (rtpmp4gpay->profile);
if (code == VOS_STARTCODE) {
/* get profile */
rtpmp4gpay->profile = g_strdup_printf ("%d", (gint) data[4]);
} else {
GST_ELEMENT_WARNING (rtpmp4gpay, STREAM, FORMAT,
(NULL), ("profile not found in config string, assuming \'1\'"));
rtpmp4gpay->profile = g_strdup ("1");
}
/* fixed rate */
rtpmp4gpay->rate = 90000;
/* video stream type */
rtpmp4gpay->streamtype = "4";
/* no params for video */
rtpmp4gpay->params = NULL;
/* mode */
rtpmp4gpay->mode = "generic";
GST_LOG_OBJECT (rtpmp4gpay, "profile %s", rtpmp4gpay->profile);
gst_buffer_unmap (buffer, data, -1);
return TRUE;
/* ERROR */
too_short:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
(NULL), ("config string too short"));
gst_buffer_unmap (buffer, data, -1);
return FALSE;
}
}
static gboolean
gst_rtp_mp4g_pay_new_caps (GstRtpMP4GPay * rtpmp4gpay)
{
gchar *config;
GValue v = { 0 };
gboolean res;
#define MP4GCAPS \
"streamtype", G_TYPE_STRING, rtpmp4gpay->streamtype, \
"profile-level-id", G_TYPE_STRING, rtpmp4gpay->profile, \
"mode", G_TYPE_STRING, rtpmp4gpay->mode, \
"config", G_TYPE_STRING, config, \
"sizelength", G_TYPE_STRING, "13", \
"indexlength", G_TYPE_STRING, "3", \
"indexdeltalength", G_TYPE_STRING, "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) {
res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
"encoding-params", G_TYPE_STRING, rtpmp4gpay->params, MP4GCAPS);
} else {
res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
MP4GCAPS);
}
g_value_unset (&v);
g_free (config);
#undef MP4GCAPS
return res;
}
static gboolean
gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
GstRtpMP4GPay *rtpmp4gpay;
GstStructure *structure;
const GValue *codec_data;
const gchar *media_type = NULL;
gboolean res;
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);
media_type = "audio";
} else if (!strcmp (name, "video/mpeg")) {
res = gst_rtp_mp4g_pay_parse_video_config (rtpmp4gpay, buffer);
media_type = "video";
} 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);
}
}
if (media_type == NULL)
goto config_failed;
gst_basertppayload_set_options (payload, media_type, TRUE, "MPEG4-GENERIC",
rtpmp4gpay->rate);
res = 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;
guint mtu;
/* 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;
guint payload_len;
guint packet_len;
GstRTPBuffer rtp;
/* 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);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
/* copy payload */
payload = gst_rtp_buffer_get_payload (&rtp);
/* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
* |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 */
/* copy stuff from adapter to payload */
gst_adapter_copy (rtpmp4gpay->adapter, &payload[4], 0, payload_len);
gst_adapter_flush (rtpmp4gpay->adapter, payload_len);
/* marker only if the packet is complete */
gst_rtp_buffer_set_marker (&rtp, avail <= payload_len);
gst_rtp_buffer_unmap (&rtp);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4gpay->first_timestamp;
GST_BUFFER_DURATION (outbuf) = rtpmp4gpay->first_duration;
if (rtpmp4gpay->frame_len) {
GST_BUFFER_OFFSET (outbuf) = rtpmp4gpay->offset;
rtpmp4gpay->offset += rtpmp4gpay->frame_len;
}
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4gpay), outbuf);
avail -= payload_len;
}
return ret;
}
/* we expect buffers as exactly one complete AU
*/
static GstFlowReturn
gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpMP4GPay *rtpmp4gpay;
rtpmp4gpay = GST_RTP_MP4G_PAY (basepayload);
rtpmp4gpay->first_timestamp = GST_BUFFER_TIMESTAMP (buffer);
rtpmp4gpay->first_duration = GST_BUFFER_DURATION (buffer);
/* we always encode and flush a full AU */
gst_adapter_push (rtpmp4gpay->adapter, buffer);
return gst_rtp_mp4g_pay_flush (rtpmp4gpay);
}
static gboolean
gst_rtp_mp4g_pay_handle_event (GstBaseRTPPayload * payload, GstEvent * event)
{
GstRtpMP4GPay *rtpmp4gpay;
rtpmp4gpay = GST_RTP_MP4G_PAY (payload);
GST_DEBUG ("Got event: %s", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:
case GST_EVENT_EOS:
/* This flush call makes sure that the last buffer is always pushed
* to the base payloader */
gst_rtp_mp4g_pay_flush (rtpmp4gpay);
break;
case GST_EVENT_FLUSH_STOP:
gst_rtp_mp4g_pay_reset (rtpmp4gpay);
break;
default:
break;
}
/* let parent handle event too */
return FALSE;
}
static GstStateChangeReturn
gst_rtp_mp4g_pay_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstRtpMP4GPay *rtpmp4gpay;
rtpmp4gpay = GST_RTP_MP4G_PAY (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_rtp_mp4g_pay_cleanup (rtpmp4gpay);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtp_mp4g_pay_cleanup (rtpmp4gpay);
break;
default:
break;
}
return ret;
}
gboolean
gst_rtp_mp4g_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpmp4gpay",
GST_RANK_SECONDARY, GST_TYPE_RTP_MP4G_PAY);
}
Reference in a new issue View git blame Copy permalink