gstreamer/gst/rtp/gstrtpspeexpay.c
Olivier Crête 7effe918d1 rtp*pay: Allocate using the base class for audio codecs
This is required to add RTP header extensions from the
meta automatically.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/674>
2020-07-17 16:53:40 -04:00

350 lines
9.4 KiB
C

/* GStreamer
* Copyright (C) <2005> Edgard Lima <edgard.lima@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 <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "gstrtpspeexpay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpspeexpay_debug);
#define GST_CAT_DEFAULT (rtpspeexpay_debug)
static GstStaticPadTemplate gst_rtp_speex_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-speex, "
"rate = (int) [ 6000, 48000 ], " "channels = (int) 1")
);
static GstStaticPadTemplate gst_rtp_speex_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) [ 6000, 48000 ], "
"encoding-name = (string) \"SPEEX\", "
"encoding-params = (string) \"1\"")
);
static GstStateChangeReturn gst_rtp_speex_pay_change_state (GstElement *
element, GstStateChange transition);
static gboolean gst_rtp_speex_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
static GstCaps *gst_rtp_speex_pay_getcaps (GstRTPBasePayload * payload,
GstPad * pad, GstCaps * filter);
static GstFlowReturn gst_rtp_speex_pay_handle_buffer (GstRTPBasePayload *
payload, GstBuffer * buffer);
#define gst_rtp_speex_pay_parent_class parent_class
G_DEFINE_TYPE (GstRtpSPEEXPay, gst_rtp_speex_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static void
gst_rtp_speex_pay_class_init (GstRtpSPEEXPayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
gstelement_class = (GstElementClass *) klass;
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gstelement_class->change_state = gst_rtp_speex_pay_change_state;
gstrtpbasepayload_class->set_caps = gst_rtp_speex_pay_setcaps;
gstrtpbasepayload_class->get_caps = gst_rtp_speex_pay_getcaps;
gstrtpbasepayload_class->handle_buffer = gst_rtp_speex_pay_handle_buffer;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_speex_pay_sink_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_speex_pay_src_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP Speex payloader", "Codec/Payloader/Network/RTP",
"Payload-encodes Speex audio into a RTP packet",
"Edgard Lima <edgard.lima@gmail.com>");
GST_DEBUG_CATEGORY_INIT (rtpspeexpay_debug, "rtpspeexpay", 0,
"Speex RTP Payloader");
}
static void
gst_rtp_speex_pay_init (GstRtpSPEEXPay * rtpspeexpay)
{
GST_RTP_BASE_PAYLOAD (rtpspeexpay)->clock_rate = 8000;
GST_RTP_BASE_PAYLOAD_PT (rtpspeexpay) = 110; /* Create String */
}
static gboolean
gst_rtp_speex_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
/* don't configure yet, we wait for the ident packet */
return TRUE;
}
static GstCaps *
gst_rtp_speex_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad,
GstCaps * filter)
{
GstCaps *otherpadcaps;
GstCaps *caps;
otherpadcaps = gst_pad_get_allowed_caps (payload->srcpad);
caps = gst_pad_get_pad_template_caps (pad);
if (otherpadcaps) {
if (!gst_caps_is_empty (otherpadcaps)) {
GstStructure *ps;
GstStructure *s;
gint clock_rate;
ps = gst_caps_get_structure (otherpadcaps, 0);
caps = gst_caps_make_writable (caps);
s = gst_caps_get_structure (caps, 0);
if (gst_structure_get_int (ps, "clock-rate", &clock_rate)) {
gst_structure_fixate_field_nearest_int (s, "rate", clock_rate);
}
}
gst_caps_unref (otherpadcaps);
}
if (filter) {
GstCaps *tcaps = caps;
caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (tcaps);
}
return caps;
}
static gboolean
gst_rtp_speex_pay_parse_ident (GstRtpSPEEXPay * rtpspeexpay,
const guint8 * data, guint size)
{
guint32 version, header_size, rate, mode, nb_channels;
GstRTPBasePayload *payload;
gchar *cstr;
gboolean res;
/* we need the header string (8), the version string (20), the version
* and the header length. */
if (size < 36)
goto too_small;
if (!g_str_has_prefix ((const gchar *) data, "Speex "))
goto wrong_header;
/* skip header and version string */
data += 28;
version = GST_READ_UINT32_LE (data);
if (version != 1)
goto wrong_version;
data += 4;
/* ensure sizes */
header_size = GST_READ_UINT32_LE (data);
if (header_size < 80)
goto header_too_small;
if (size < header_size)
goto payload_too_small;
data += 4;
rate = GST_READ_UINT32_LE (data);
data += 4;
mode = GST_READ_UINT32_LE (data);
data += 8;
nb_channels = GST_READ_UINT32_LE (data);
GST_DEBUG_OBJECT (rtpspeexpay, "rate %d, mode %d, nb_channels %d",
rate, mode, nb_channels);
payload = GST_RTP_BASE_PAYLOAD (rtpspeexpay);
gst_rtp_base_payload_set_options (payload, "audio", FALSE, "SPEEX", rate);
cstr = g_strdup_printf ("%d", nb_channels);
res = gst_rtp_base_payload_set_outcaps (payload, "encoding-params",
G_TYPE_STRING, cstr, NULL);
g_free (cstr);
return res;
/* ERRORS */
too_small:
{
GST_DEBUG_OBJECT (rtpspeexpay,
"ident packet too small, need at least 32 bytes");
return FALSE;
}
wrong_header:
{
GST_DEBUG_OBJECT (rtpspeexpay,
"ident packet does not start with \"Speex \"");
return FALSE;
}
wrong_version:
{
GST_DEBUG_OBJECT (rtpspeexpay, "can only handle version 1, have version %d",
version);
return FALSE;
}
header_too_small:
{
GST_DEBUG_OBJECT (rtpspeexpay,
"header size too small, need at least 80 bytes, " "got only %d",
header_size);
return FALSE;
}
payload_too_small:
{
GST_DEBUG_OBJECT (rtpspeexpay,
"payload too small, need at least %d bytes, got only %d", header_size,
size);
return FALSE;
}
}
static GstFlowReturn
gst_rtp_speex_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpSPEEXPay *rtpspeexpay;
GstMapInfo map;
GstBuffer *outbuf;
GstClockTime timestamp, duration;
GstFlowReturn ret;
rtpspeexpay = GST_RTP_SPEEX_PAY (basepayload);
gst_buffer_map (buffer, &map, GST_MAP_READ);
switch (rtpspeexpay->packet) {
case 0:
/* ident packet. We need to parse the headers to construct the RTP
* properties. */
if (!gst_rtp_speex_pay_parse_ident (rtpspeexpay, map.data, map.size)) {
gst_buffer_unmap (buffer, &map);
goto parse_error;
}
ret = GST_FLOW_OK;
gst_buffer_unmap (buffer, &map);
goto done;
case 1:
/* comment packet, we ignore it */
ret = GST_FLOW_OK;
gst_buffer_unmap (buffer, &map);
goto done;
default:
/* other packets go in the payload */
break;
}
gst_buffer_unmap (buffer, &map);
if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_GAP)) {
ret = GST_FLOW_OK;
goto done;
}
timestamp = GST_BUFFER_PTS (buffer);
duration = GST_BUFFER_DURATION (buffer);
/* FIXME, only one SPEEX frame per RTP packet for now */
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
/* FIXME, assert for now */
g_assert (gst_buffer_get_size (buffer) <=
GST_RTP_BASE_PAYLOAD_MTU (rtpspeexpay));
/* copy timestamp and duration */
GST_BUFFER_PTS (outbuf) = timestamp;
GST_BUFFER_DURATION (outbuf) = duration;
gst_rtp_copy_audio_meta (basepayload, outbuf, buffer);
outbuf = gst_buffer_append (outbuf, buffer);
buffer = NULL;
ret = gst_rtp_base_payload_push (basepayload, outbuf);
done:
if (buffer)
gst_buffer_unref (buffer);
rtpspeexpay->packet++;
return ret;
/* ERRORS */
parse_error:
{
GST_ELEMENT_ERROR (rtpspeexpay, STREAM, DECODE, (NULL),
("Error parsing first identification packet."));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
}
static GstStateChangeReturn
gst_rtp_speex_pay_change_state (GstElement * element, GstStateChange transition)
{
GstRtpSPEEXPay *rtpspeexpay;
GstStateChangeReturn ret;
rtpspeexpay = GST_RTP_SPEEX_PAY (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
rtpspeexpay->packet = 0;
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
gboolean
gst_rtp_speex_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpspeexpay",
GST_RANK_SECONDARY, GST_TYPE_RTP_SPEEX_PAY);
}