mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-09 13:48:55 +00:00
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>
504 lines
14 KiB
C
504 lines
14 KiB
C
/* GStreamer
|
|
* Copyright (C) <2009> 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., 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 "gstrtpceltpay.h"
|
|
#include "gstrtputils.h"
|
|
|
|
GST_DEBUG_CATEGORY_STATIC (rtpceltpay_debug);
|
|
#define GST_CAT_DEFAULT (rtpceltpay_debug)
|
|
|
|
static GstStaticPadTemplate gst_rtp_celt_pay_sink_template =
|
|
GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-celt, "
|
|
"rate = (int) [ 32000, 64000 ], "
|
|
"channels = (int) [1, 2], " "frame-size = (int) [ 64, 512 ]")
|
|
);
|
|
|
|
static GstStaticPadTemplate gst_rtp_celt_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) [ 32000, 48000 ], "
|
|
"encoding-name = (string) \"CELT\"")
|
|
);
|
|
|
|
static void gst_rtp_celt_pay_finalize (GObject * object);
|
|
|
|
static GstStateChangeReturn gst_rtp_celt_pay_change_state (GstElement *
|
|
element, GstStateChange transition);
|
|
|
|
static gboolean gst_rtp_celt_pay_setcaps (GstRTPBasePayload * payload,
|
|
GstCaps * caps);
|
|
static GstCaps *gst_rtp_celt_pay_getcaps (GstRTPBasePayload * payload,
|
|
GstPad * pad, GstCaps * filter);
|
|
static GstFlowReturn gst_rtp_celt_pay_handle_buffer (GstRTPBasePayload *
|
|
payload, GstBuffer * buffer);
|
|
|
|
#define gst_rtp_celt_pay_parent_class parent_class
|
|
G_DEFINE_TYPE (GstRtpCELTPay, gst_rtp_celt_pay, GST_TYPE_RTP_BASE_PAYLOAD);
|
|
|
|
static void
|
|
gst_rtp_celt_pay_class_init (GstRtpCELTPayClass * klass)
|
|
{
|
|
GObjectClass *gobject_class;
|
|
GstElementClass *gstelement_class;
|
|
GstRTPBasePayloadClass *gstrtpbasepayload_class;
|
|
|
|
GST_DEBUG_CATEGORY_INIT (rtpceltpay_debug, "rtpceltpay", 0,
|
|
"CELT RTP Payloader");
|
|
|
|
gobject_class = (GObjectClass *) klass;
|
|
gstelement_class = (GstElementClass *) klass;
|
|
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
|
|
|
|
gobject_class->finalize = gst_rtp_celt_pay_finalize;
|
|
|
|
gstelement_class->change_state = gst_rtp_celt_pay_change_state;
|
|
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&gst_rtp_celt_pay_sink_template);
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&gst_rtp_celt_pay_src_template);
|
|
|
|
gst_element_class_set_static_metadata (gstelement_class, "RTP CELT payloader",
|
|
"Codec/Payloader/Network/RTP",
|
|
"Payload-encodes CELT audio into a RTP packet",
|
|
"Wim Taymans <wim.taymans@gmail.com>");
|
|
|
|
gstrtpbasepayload_class->set_caps = gst_rtp_celt_pay_setcaps;
|
|
gstrtpbasepayload_class->get_caps = gst_rtp_celt_pay_getcaps;
|
|
gstrtpbasepayload_class->handle_buffer = gst_rtp_celt_pay_handle_buffer;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_celt_pay_init (GstRtpCELTPay * rtpceltpay)
|
|
{
|
|
rtpceltpay->queue = g_queue_new ();
|
|
}
|
|
|
|
static void
|
|
gst_rtp_celt_pay_finalize (GObject * object)
|
|
{
|
|
GstRtpCELTPay *rtpceltpay;
|
|
|
|
rtpceltpay = GST_RTP_CELT_PAY (object);
|
|
|
|
g_queue_free (rtpceltpay->queue);
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_celt_pay_clear_queued (GstRtpCELTPay * rtpceltpay)
|
|
{
|
|
GstBuffer *buf;
|
|
|
|
while ((buf = g_queue_pop_head (rtpceltpay->queue)))
|
|
gst_buffer_unref (buf);
|
|
|
|
rtpceltpay->bytes = 0;
|
|
rtpceltpay->sbytes = 0;
|
|
rtpceltpay->qduration = 0;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_celt_pay_add_queued (GstRtpCELTPay * rtpceltpay, GstBuffer * buffer,
|
|
guint ssize, guint size, GstClockTime duration)
|
|
{
|
|
g_queue_push_tail (rtpceltpay->queue, buffer);
|
|
rtpceltpay->sbytes += ssize;
|
|
rtpceltpay->bytes += size;
|
|
/* only add durations when we have a valid previous duration */
|
|
if (rtpceltpay->qduration != -1) {
|
|
if (duration != -1)
|
|
/* only add valid durations */
|
|
rtpceltpay->qduration += duration;
|
|
else
|
|
/* if we add a buffer without valid duration, our total queued duration
|
|
* becomes unknown */
|
|
rtpceltpay->qduration = -1;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_celt_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
|
|
{
|
|
/* don't configure yet, we wait for the ident packet */
|
|
return TRUE;
|
|
}
|
|
|
|
|
|
static GstCaps *
|
|
gst_rtp_celt_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad,
|
|
GstCaps * filter)
|
|
{
|
|
GstCaps *otherpadcaps;
|
|
GstCaps *caps;
|
|
const gchar *params;
|
|
|
|
caps = gst_pad_get_pad_template_caps (pad);
|
|
|
|
otherpadcaps = gst_pad_get_allowed_caps (payload->srcpad);
|
|
if (otherpadcaps) {
|
|
if (!gst_caps_is_empty (otherpadcaps)) {
|
|
GstStructure *ps;
|
|
GstStructure *s;
|
|
gint clock_rate = 0, frame_size = 0, channels = 1;
|
|
|
|
caps = gst_caps_make_writable (caps);
|
|
|
|
ps = gst_caps_get_structure (otherpadcaps, 0);
|
|
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);
|
|
}
|
|
|
|
if ((params = gst_structure_get_string (ps, "frame-size")))
|
|
frame_size = atoi (params);
|
|
if (frame_size)
|
|
gst_structure_set (s, "frame-size", G_TYPE_INT, frame_size, NULL);
|
|
|
|
if ((params = gst_structure_get_string (ps, "encoding-params"))) {
|
|
channels = atoi (params);
|
|
gst_structure_fixate_field_nearest_int (s, "channels", channels);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (payload, "clock-rate=%d frame-size=%d channels=%d",
|
|
clock_rate, frame_size, channels);
|
|
}
|
|
gst_caps_unref (otherpadcaps);
|
|
}
|
|
|
|
if (filter) {
|
|
GstCaps *tmp;
|
|
|
|
GST_DEBUG_OBJECT (payload, "Intersect %" GST_PTR_FORMAT " and filter %"
|
|
GST_PTR_FORMAT, caps, filter);
|
|
tmp = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
|
|
gst_caps_unref (caps);
|
|
caps = tmp;
|
|
}
|
|
|
|
return caps;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_celt_pay_parse_ident (GstRtpCELTPay * rtpceltpay,
|
|
const guint8 * data, guint size)
|
|
{
|
|
guint32 version, header_size, rate, nb_channels, frame_size, overlap;
|
|
guint32 bytes_per_packet;
|
|
GstRTPBasePayload *payload;
|
|
gchar *cstr, *fsstr;
|
|
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, "CELT "))
|
|
goto wrong_header;
|
|
|
|
/* skip header and version string */
|
|
data += 28;
|
|
|
|
version = GST_READ_UINT32_LE (data);
|
|
GST_DEBUG_OBJECT (rtpceltpay, "version %08x", version);
|
|
#if 0
|
|
if (version != 1)
|
|
goto wrong_version;
|
|
#endif
|
|
|
|
data += 4;
|
|
/* ensure sizes */
|
|
header_size = GST_READ_UINT32_LE (data);
|
|
if (header_size < 56)
|
|
goto header_too_small;
|
|
|
|
if (size < header_size)
|
|
goto payload_too_small;
|
|
|
|
data += 4;
|
|
rate = GST_READ_UINT32_LE (data);
|
|
data += 4;
|
|
nb_channels = GST_READ_UINT32_LE (data);
|
|
data += 4;
|
|
frame_size = GST_READ_UINT32_LE (data);
|
|
data += 4;
|
|
overlap = GST_READ_UINT32_LE (data);
|
|
data += 4;
|
|
bytes_per_packet = GST_READ_UINT32_LE (data);
|
|
|
|
GST_DEBUG_OBJECT (rtpceltpay, "rate %d, nb_channels %d, frame_size %d",
|
|
rate, nb_channels, frame_size);
|
|
GST_DEBUG_OBJECT (rtpceltpay, "overlap %d, bytes_per_packet %d",
|
|
overlap, bytes_per_packet);
|
|
|
|
payload = GST_RTP_BASE_PAYLOAD (rtpceltpay);
|
|
|
|
gst_rtp_base_payload_set_options (payload, "audio", FALSE, "CELT", rate);
|
|
cstr = g_strdup_printf ("%d", nb_channels);
|
|
fsstr = g_strdup_printf ("%d", frame_size);
|
|
res = gst_rtp_base_payload_set_outcaps (payload, "encoding-params",
|
|
G_TYPE_STRING, cstr, "frame-size", G_TYPE_STRING, fsstr, NULL);
|
|
g_free (cstr);
|
|
g_free (fsstr);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
too_small:
|
|
{
|
|
GST_DEBUG_OBJECT (rtpceltpay,
|
|
"ident packet too small, need at least 32 bytes");
|
|
return FALSE;
|
|
}
|
|
wrong_header:
|
|
{
|
|
GST_DEBUG_OBJECT (rtpceltpay,
|
|
"ident packet does not start with \"CELT \"");
|
|
return FALSE;
|
|
}
|
|
#if 0
|
|
wrong_version:
|
|
{
|
|
GST_DEBUG_OBJECT (rtpceltpay, "can only handle version 1, have version %d",
|
|
version);
|
|
return FALSE;
|
|
}
|
|
#endif
|
|
header_too_small:
|
|
{
|
|
GST_DEBUG_OBJECT (rtpceltpay,
|
|
"header size too small, need at least 80 bytes, " "got only %d",
|
|
header_size);
|
|
return FALSE;
|
|
}
|
|
payload_too_small:
|
|
{
|
|
GST_DEBUG_OBJECT (rtpceltpay,
|
|
"payload too small, need at least %d bytes, got only %d", header_size,
|
|
size);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_celt_pay_flush_queued (GstRtpCELTPay * rtpceltpay)
|
|
{
|
|
GstFlowReturn ret;
|
|
GstBuffer *buf, *outbuf;
|
|
guint8 *payload, *spayload;
|
|
guint payload_len;
|
|
GstClockTime duration;
|
|
GstRTPBuffer rtp = { NULL, };
|
|
|
|
payload_len = rtpceltpay->bytes + rtpceltpay->sbytes;
|
|
duration = rtpceltpay->qduration;
|
|
|
|
GST_DEBUG_OBJECT (rtpceltpay, "flushing out %u, duration %" GST_TIME_FORMAT,
|
|
payload_len, GST_TIME_ARGS (rtpceltpay->qduration));
|
|
|
|
/* get a big enough packet for the sizes + payloads */
|
|
outbuf =
|
|
gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
|
|
(rtpceltpay), payload_len, 0, 0);
|
|
|
|
GST_BUFFER_DURATION (outbuf) = duration;
|
|
|
|
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
|
|
|
|
/* point to the payload for size headers and data */
|
|
spayload = gst_rtp_buffer_get_payload (&rtp);
|
|
payload = spayload + rtpceltpay->sbytes;
|
|
|
|
while ((buf = g_queue_pop_head (rtpceltpay->queue))) {
|
|
guint size;
|
|
|
|
/* copy first timestamp to output */
|
|
if (GST_BUFFER_PTS (outbuf) == -1)
|
|
GST_BUFFER_PTS (outbuf) = GST_BUFFER_PTS (buf);
|
|
|
|
/* write the size to the header */
|
|
size = gst_buffer_get_size (buf);
|
|
while (size > 0xff) {
|
|
*spayload++ = 0xff;
|
|
size -= 0xff;
|
|
}
|
|
*spayload++ = size;
|
|
|
|
/* copy payload */
|
|
size = gst_buffer_get_size (buf);
|
|
gst_buffer_extract (buf, 0, payload, size);
|
|
payload += size;
|
|
|
|
gst_rtp_copy_audio_meta (rtpceltpay, outbuf, buf);
|
|
|
|
gst_buffer_unref (buf);
|
|
}
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
/* we consumed it all */
|
|
rtpceltpay->bytes = 0;
|
|
rtpceltpay->sbytes = 0;
|
|
rtpceltpay->qduration = 0;
|
|
|
|
ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpceltpay), outbuf);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_celt_pay_handle_buffer (GstRTPBasePayload * basepayload,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstFlowReturn ret;
|
|
GstRtpCELTPay *rtpceltpay;
|
|
gsize payload_len;
|
|
GstMapInfo map;
|
|
GstClockTime duration, packet_dur;
|
|
guint i, ssize, packet_len;
|
|
|
|
rtpceltpay = GST_RTP_CELT_PAY (basepayload);
|
|
|
|
ret = GST_FLOW_OK;
|
|
|
|
gst_buffer_map (buffer, &map, GST_MAP_READ);
|
|
|
|
switch (rtpceltpay->packet) {
|
|
case 0:
|
|
/* ident packet. We need to parse the headers to construct the RTP
|
|
* properties. */
|
|
if (!gst_rtp_celt_pay_parse_ident (rtpceltpay, map.data, map.size))
|
|
goto parse_error;
|
|
|
|
goto cleanup;
|
|
case 1:
|
|
/* comment packet, we ignore it */
|
|
goto cleanup;
|
|
default:
|
|
/* other packets go in the payload */
|
|
break;
|
|
}
|
|
gst_buffer_unmap (buffer, &map);
|
|
|
|
duration = GST_BUFFER_DURATION (buffer);
|
|
|
|
GST_LOG_OBJECT (rtpceltpay,
|
|
"got buffer of duration %" GST_TIME_FORMAT ", size %" G_GSIZE_FORMAT,
|
|
GST_TIME_ARGS (duration), map.size);
|
|
|
|
/* calculate the size of the size field and the payload */
|
|
ssize = 1;
|
|
for (i = map.size; i > 0xff; i -= 0xff)
|
|
ssize++;
|
|
|
|
GST_DEBUG_OBJECT (rtpceltpay, "bytes for size %u", ssize);
|
|
|
|
/* calculate what the new size and duration would be of the packet */
|
|
payload_len = ssize + map.size + rtpceltpay->bytes + rtpceltpay->sbytes;
|
|
if (rtpceltpay->qduration != -1 && duration != -1)
|
|
packet_dur = rtpceltpay->qduration + duration;
|
|
else
|
|
packet_dur = 0;
|
|
|
|
packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
|
|
|
|
if (gst_rtp_base_payload_is_filled (basepayload, packet_len, packet_dur)) {
|
|
/* size or duration would overflow the packet, flush the queued data */
|
|
ret = gst_rtp_celt_pay_flush_queued (rtpceltpay);
|
|
}
|
|
|
|
/* queue the packet */
|
|
gst_rtp_celt_pay_add_queued (rtpceltpay, buffer, ssize, map.size, duration);
|
|
|
|
done:
|
|
rtpceltpay->packet++;
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
cleanup:
|
|
{
|
|
gst_buffer_unmap (buffer, &map);
|
|
goto done;
|
|
}
|
|
parse_error:
|
|
{
|
|
GST_ELEMENT_ERROR (rtpceltpay, STREAM, DECODE, (NULL),
|
|
("Error parsing first identification packet."));
|
|
gst_buffer_unmap (buffer, &map);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_celt_pay_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstRtpCELTPay *rtpceltpay;
|
|
GstStateChangeReturn ret;
|
|
|
|
rtpceltpay = GST_RTP_CELT_PAY (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
rtpceltpay->packet = 0;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_rtp_celt_pay_clear_queued (rtpceltpay);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_celt_pay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtpceltpay",
|
|
GST_RANK_SECONDARY, GST_TYPE_RTP_CELT_PAY);
|
|
}
|