gstreamer/gst/rtp/gstrtpopuspay.c
Havard Graff d9aaa15a30 rtpopuspay: make depay ! pay work
There is a use-case for a server to re-payload opus going through it.

Problem was that the payloader requires channels in the caps, but
this is not something the depayloader can parse out of the stream, meaning
caps-negotiation would fail.

Removing the requirement of channels in the template-caps fixes this.
2020-04-03 09:04:32 +00:00

261 lines
7.8 KiB
C

/*
* Opus Payloader 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 <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "gstrtpopuspay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpopuspay_debug);
#define GST_CAT_DEFAULT (rtpopuspay_debug)
static GstStaticPadTemplate gst_rtp_opus_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-opus, channel-mapping-family = (int) 0")
);
static GstStaticPadTemplate gst_rtp_opus_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) 48000, "
"encoding-params = (string) \"2\", "
"encoding-name = (string) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\" }")
);
static gboolean gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
static GstCaps *gst_rtp_opus_pay_getcaps (GstRTPBasePayload * payload,
GstPad * pad, GstCaps * filter);
static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload *
payload, GstBuffer * buffer);
G_DEFINE_TYPE (GstRtpOPUSPay, gst_rtp_opus_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static void
gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
{
GstRTPBasePayloadClass *gstbasertppayload_class;
GstElementClass *element_class;
gstbasertppayload_class = (GstRTPBasePayloadClass *) klass;
element_class = GST_ELEMENT_CLASS (klass);
gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps;
gstbasertppayload_class->get_caps = gst_rtp_opus_pay_getcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer;
gst_element_class_add_static_pad_template (element_class,
&gst_rtp_opus_pay_src_template);
gst_element_class_add_static_pad_template (element_class,
&gst_rtp_opus_pay_sink_template);
gst_element_class_set_static_metadata (element_class,
"RTP Opus payloader",
"Codec/Payloader/Network/RTP",
"Puts Opus audio in RTP packets",
"Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
GST_DEBUG_CATEGORY_INIT (rtpopuspay_debug, "rtpopuspay", 0,
"Opus RTP Payloader");
}
static void
gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay)
{
}
static gboolean
gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
GstCaps *src_caps;
GstStructure *s;
const char *encoding_name = "OPUS";
gint channels, rate;
const char *sprop_stereo = NULL;
char *sprop_maxcapturerate = NULL;
src_caps = gst_pad_get_allowed_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));
if (src_caps) {
GstStructure *s;
const GValue *value;
s = gst_caps_get_structure (src_caps, 0);
if (gst_structure_has_field (s, "encoding-name")) {
GValue default_value = G_VALUE_INIT;
g_value_init (&default_value, G_TYPE_STRING);
g_value_set_static_string (&default_value, encoding_name);
value = gst_structure_get_value (s, "encoding-name");
if (!gst_value_can_intersect (&default_value, value))
encoding_name = "X-GST-OPUS-DRAFT-SPITTKA-00";
}
gst_caps_unref (src_caps);
}
s = gst_caps_get_structure (caps, 0);
if (gst_structure_get_int (s, "channels", &channels)) {
if (channels > 2) {
GST_ERROR_OBJECT (payload,
"More than 2 channels with channel-mapping-family=0 is invalid");
return FALSE;
} else if (channels == 2) {
sprop_stereo = "1";
} else {
sprop_stereo = "0";
}
}
if (gst_structure_get_int (s, "rate", &rate)) {
sprop_maxcapturerate = g_strdup_printf ("%d", rate);
}
gst_rtp_base_payload_set_options (payload, "audio", FALSE,
encoding_name, 48000);
if (sprop_maxcapturerate && sprop_stereo) {
res =
gst_rtp_base_payload_set_outcaps (payload, "sprop-maxcapturerate",
G_TYPE_STRING, sprop_maxcapturerate, "sprop-stereo", G_TYPE_STRING,
sprop_stereo, NULL);
} else if (sprop_maxcapturerate) {
res =
gst_rtp_base_payload_set_outcaps (payload, "sprop-maxcapturerate",
G_TYPE_STRING, sprop_maxcapturerate, NULL);
} else if (sprop_stereo) {
res =
gst_rtp_base_payload_set_outcaps (payload, "sprop-stereo",
G_TYPE_STRING, sprop_stereo, NULL);
} else {
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
}
g_free (sprop_maxcapturerate);
return res;
}
static GstFlowReturn
gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstBuffer *outbuf;
GstClockTime pts, dts, duration;
pts = GST_BUFFER_PTS (buffer);
dts = GST_BUFFER_DTS (buffer);
duration = GST_BUFFER_DURATION (buffer);
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
gst_rtp_copy_audio_meta (basepayload, outbuf, buffer);
outbuf = gst_buffer_append (outbuf, buffer);
GST_BUFFER_PTS (outbuf) = pts;
GST_BUFFER_DTS (outbuf) = dts;
GST_BUFFER_DURATION (outbuf) = duration;
/* Push out */
return gst_rtp_base_payload_push (basepayload, outbuf);
}
static GstCaps *
gst_rtp_opus_pay_getcaps (GstRTPBasePayload * payload,
GstPad * pad, GstCaps * filter)
{
GstCaps *caps, *peercaps, *tcaps;
GstStructure *s;
const gchar *stereo;
if (pad == GST_RTP_BASE_PAYLOAD_SRCPAD (payload))
return
GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps
(payload, pad, filter);
tcaps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));
peercaps = gst_pad_peer_query_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload),
tcaps);
gst_caps_unref (tcaps);
if (!peercaps)
return
GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps
(payload, pad, filter);
if (gst_caps_is_empty (peercaps))
return peercaps;
caps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SINKPAD (payload));
s = gst_caps_get_structure (peercaps, 0);
stereo = gst_structure_get_string (s, "stereo");
if (stereo != NULL) {
caps = gst_caps_make_writable (caps);
if (!strcmp (stereo, "1")) {
GstCaps *caps2 = gst_caps_copy (caps);
gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL);
gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 1, NULL);
caps = gst_caps_merge (caps, caps2);
} else if (!strcmp (stereo, "0")) {
GstCaps *caps2 = gst_caps_copy (caps);
gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 2, NULL);
caps = gst_caps_merge (caps, caps2);
}
}
gst_caps_unref (peercaps);
if (filter) {
GstCaps *tmp = gst_caps_intersect_full (caps, filter,
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = tmp;
}
GST_DEBUG_OBJECT (payload, "Returning caps: %" GST_PTR_FORMAT, caps);
return caps;
}
gboolean
gst_rtp_opus_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpopuspay",
GST_RANK_PRIMARY, GST_TYPE_RTP_OPUS_PAY);
}