mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-30 04:00:37 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/rtp/gstrtpmp4gpay.c
Robert Swain 5b18c652fb rtp, rtpmanager: Address unused but set variables 
GCC 4.6.x spits warnings about variables that are unused but set. Such
variables have been removed where trivial but with comments left behind
for informational purposes in some cases.

gst_rtp_session_chain_recv_rtcp () was changed in commit 490113d4
to always return GST_FLOW_OK instead of the return value of
rtp_session_process_rtcp (), so we'll keep it that way.
2011-04-16 12:49:16 +01:00

631 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 (GstPad * pad, GstEvent * event);
GST_BOILERPLATE (GstRtpMP4GPay, gst_rtp_mp4g_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD)
static void gst_rtp_mp4g_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_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_simple (element_class, "RTP MPEG4 ES payloader",
"Codec/Payloader/Network/RTP",
"Payload MPEG4 elementary streams as RTP packets (RFC 3640)",
"Wim Taymans <wim.taymans@gmail.com>");
}
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_DEBUG_CATEGORY_INIT (rtpmp4gpay_debug, "rtpmp4gpay", 0,
"MP4-generic RTP Payloader");
}
static void
gst_rtp_mp4g_pay_init (GstRtpMP4GPay * rtpmp4gpay, GstRtpMP4GPayClass * klass)
{
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;
guint size;
guint8 objectType = 0;
guint8 samplingIdx = 0;
guint8 channelCfg = 0;
GstBitReader br;
data = GST_BUFFER_DATA (buffer);
size = GST_BUFFER_SIZE (buffer);
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);
return TRUE;
/* ERROR */
too_short:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
(NULL), ("config string too short"));
return FALSE;
}
invalid_object:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
(NULL), ("invalid object type"));
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, must < 8", 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);
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);
return TRUE;
/* ERROR */
too_short:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
(NULL), ("config string too short"));
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;
/* 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 */
/* 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 (outbuf, avail <= payload_len);
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 (GstPad * pad, GstEvent * event)
{
GstRtpMP4GPay *rtpmp4gpay;
rtpmp4gpay = GST_RTP_MP4G_PAY (gst_pad_get_parent (pad));
GST_DEBUG ("Got event: %s", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
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;
}
g_object_unref (rtpmp4gpay);
/* 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