gstreamer/subprojects/gst-plugins-good/gst/rtp/gstrtpopusdepay.c
Olivier Crête c2a357c867 rtpopusdepay: set resync flag
- Set re-sync flag on output buffer when rtp had the marker flag set.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5529>
2023-11-10 21:45:13 +00:00

266 lines
8.7 KiB
C

/*
* Opus Depayloader Gst Element
*
* @author: Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>
*
* 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 <string.h>
#include <stdlib.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "gstrtpelements.h"
#include "gstrtpopusdepay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpopusdepay_debug);
#define GST_CAT_DEFAULT (rtpopusdepay_debug)
static GstStaticPadTemplate gst_rtp_opus_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ","
"clock-rate = (int) 48000, "
"encoding-name = (string) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\", \"MULTIOPUS\" }")
);
static GstStaticPadTemplate gst_rtp_opus_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-opus, channel-mapping-family = (int) [ 0, 1 ]")
);
static GstBuffer *gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp_buffer);
static gboolean gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
G_DEFINE_TYPE (GstRTPOpusDepay, gst_rtp_opus_depay,
GST_TYPE_RTP_BASE_DEPAYLOAD);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpopusdepay, "rtpopusdepay",
GST_RANK_PRIMARY, GST_TYPE_RTP_OPUS_DEPAY, rtp_element_init (plugin));
static void
gst_rtp_opus_depay_class_init (GstRTPOpusDepayClass * klass)
{
GstRTPBaseDepayloadClass *gstbasertpdepayload_class;
GstElementClass *element_class;
element_class = GST_ELEMENT_CLASS (klass);
gstbasertpdepayload_class = (GstRTPBaseDepayloadClass *) klass;
gst_element_class_add_static_pad_template (element_class,
&gst_rtp_opus_depay_src_template);
gst_element_class_add_static_pad_template (element_class,
&gst_rtp_opus_depay_sink_template);
gst_element_class_set_static_metadata (element_class,
"RTP Opus packet depayloader", "Codec/Depayloader/Network/RTP",
"Extracts Opus audio from RTP packets",
"Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
gstbasertpdepayload_class->process_rtp_packet = gst_rtp_opus_depay_process;
gstbasertpdepayload_class->set_caps = gst_rtp_opus_depay_setcaps;
GST_DEBUG_CATEGORY_INIT (rtpopusdepay_debug, "rtpopusdepay", 0,
"Opus RTP Depayloader");
}
static void
gst_rtp_opus_depay_init (GstRTPOpusDepay * rtpopusdepay)
{
}
static gboolean
gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstCaps *srccaps;
GstStructure *s;
gboolean ret;
const gchar *sprop_maxcapturerate;
/* Default unless overridden by sprop_maxcapturerate */
gint rate = 48000;
srccaps = gst_caps_new_empty_simple ("audio/x-opus");
s = gst_caps_get_structure (caps, 0);
if (g_str_equal (gst_structure_get_string (s, "encoding-name"), "MULTIOPUS")) {
gint channels;
gint stream_count;
gint coupled_count;
const gchar *encoding_params;
const gchar *num_streams;
const gchar *coupled_streams;
const gchar *channel_mapping;
gchar *endptr;
if (!gst_structure_has_field_typed (s, "encoding-params", G_TYPE_STRING) ||
!gst_structure_has_field_typed (s, "num_streams", G_TYPE_STRING) ||
!gst_structure_has_field_typed (s, "coupled_streams", G_TYPE_STRING) ||
!gst_structure_has_field_typed (s, "channel_mapping", G_TYPE_STRING)) {
GST_WARNING_OBJECT (depayload, "Encoding name 'MULTIOPUS' requires "
"encoding-params, num_streams, coupled_streams and channel_mapping "
"as string fields in caps.");
goto reject_caps;
}
gst_caps_set_simple (srccaps, "channel-mapping-family", G_TYPE_INT, 1,
NULL);
encoding_params = gst_structure_get_string (s, "encoding-params");
channels = g_ascii_strtoull (encoding_params, &endptr, 10);
if (*endptr != '\0' || channels > 255) {
GST_WARNING_OBJECT (depayload, "Invalid encoding-params value '%s'",
encoding_params);
goto reject_caps;
}
gst_caps_set_simple (srccaps, "channels", G_TYPE_INT, channels, NULL);
num_streams = gst_structure_get_string (s, "num_streams");
stream_count = g_ascii_strtoull (num_streams, &endptr, 10);
if (*endptr != '\0' || stream_count > channels) {
GST_WARNING_OBJECT (depayload, "Invalid num_streams value '%s'",
num_streams);
goto reject_caps;
}
gst_caps_set_simple (srccaps, "stream-count", G_TYPE_INT, stream_count,
NULL);
coupled_streams = gst_structure_get_string (s, "coupled_streams");
coupled_count = g_ascii_strtoull (coupled_streams, &endptr, 10);
if (*endptr != '\0' || coupled_count > stream_count) {
GST_WARNING_OBJECT (depayload, "Invalid coupled_streams value '%s'",
coupled_streams);
goto reject_caps;
}
gst_caps_set_simple (srccaps, "coupled-count", G_TYPE_INT, coupled_count,
NULL);
channel_mapping = gst_structure_get_string (s, "channel_mapping");
{
gchar **split;
gchar **ptr;
GValue mapping = G_VALUE_INIT;
GValue v = G_VALUE_INIT;
split = g_strsplit (channel_mapping, ",", -1);
g_value_init (&mapping, GST_TYPE_ARRAY);
g_value_init (&v, G_TYPE_INT);
for (ptr = split; *ptr; ++ptr) {
gint channel = g_ascii_strtoull (*ptr, &endptr, 10);
if (*endptr != '\0' || channel > channels) {
GST_WARNING_OBJECT (depayload, "Invalid channel_mapping value '%s'",
channel_mapping);
g_value_unset (&mapping);
break;
}
g_value_set_int (&v, channel);
gst_value_array_append_value (&mapping, &v);
}
g_value_unset (&v);
g_strfreev (split);
if (G_IS_VALUE (&mapping)) {
gst_caps_set_value (srccaps, "channel-mapping", &mapping);
g_value_unset (&mapping);
} else {
goto reject_caps;
}
}
} else {
const gchar *sprop_stereo;
gst_caps_set_simple (srccaps, "channel-mapping-family", G_TYPE_INT, 0,
NULL);
if ((sprop_stereo = gst_structure_get_string (s, "sprop-stereo"))) {
if (strcmp (sprop_stereo, "0") == 0)
gst_caps_set_simple (srccaps, "channels", G_TYPE_INT, 1, NULL);
else if (strcmp (sprop_stereo, "1") == 0)
gst_caps_set_simple (srccaps, "channels", G_TYPE_INT, 2, NULL);
else
GST_WARNING_OBJECT (depayload, "Unknown sprop-stereo value '%s'",
sprop_stereo);
} else {
/* Although sprop-stereo defaults to mono as per RFC 7587, this just means
that the signal is likely mono and can be safely downmixed, it may
still be stereo at times. */
gst_caps_set_simple (srccaps, "channels", G_TYPE_INT, 2, NULL);
}
}
if ((sprop_maxcapturerate =
gst_structure_get_string (s, "sprop-maxcapturerate"))) {
gchar *tailptr;
gulong tmp_rate;
tmp_rate = strtoul (sprop_maxcapturerate, &tailptr, 10);
if (tmp_rate > INT_MAX || *tailptr != '\0') {
GST_WARNING_OBJECT (depayload,
"Failed to parse sprop-maxcapturerate value '%s'",
sprop_maxcapturerate);
} else {
/* Valid rate from sprop, let's use it */
rate = tmp_rate;
}
}
gst_caps_set_simple (srccaps, "rate", G_TYPE_INT, rate, NULL);
ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
GST_DEBUG_OBJECT (depayload,
"set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret);
gst_caps_unref (srccaps);
depayload->clock_rate = 48000;
return ret;
reject_caps:
gst_caps_unref (srccaps);
return FALSE;
}
static GstBuffer *
gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp_buffer)
{
GstBuffer *outbuf;
outbuf = gst_rtp_buffer_get_payload_buffer (rtp_buffer);
if (gst_rtp_buffer_get_marker (rtp_buffer))
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
gst_rtp_drop_non_audio_meta (depayload, outbuf);
return outbuf;
}