rtp: add rtpisacpay

Payload for the iSAC audio codec.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/530>

diff --git a/gst/rtp/gstrtp.c b/gst/rtp/gstrtp.c
index b2a6706..f9a823f 100644 (file)
--- a/gst/rtp/gstrtp.c
+++ b/gst/rtp/gstrtp.c
@@ -70,6 +70,7 @@
 #include "gstrtph264pay.h"
 #include "gstrtph265depay.h"
 #include "gstrtph265pay.h"
+#include "gstrtpisacpay.h"
 #include "gstrtpj2kdepay.h"
 #include "gstrtpj2kpay.h"
 #include "gstrtpjpegdepay.h"
@@ -393,6 +394,9 @@ plugin_init (GstPlugin * plugin)
   if (!gst_rtp_stream_depay_plugin_init (plugin))
     return FALSE;
 
+  if (!gst_rtp_isac_pay_plugin_init (plugin))
+    return FALSE;
+
   if (!gst_element_register (plugin, "rtpredenc", GST_RANK_NONE,
           GST_TYPE_RTP_RED_ENC))
     return FALSE;

diff --git a/gst/rtp/gstrtpisacpay.c b/gst/rtp/gstrtpisacpay.c
new file mode 100644 (file)
index 0000000..750d975
--- /dev/null
+++ b/gst/rtp/gstrtpisacpay.c
@@ -0,0 +1,187 @@
+/* GStreamer
+ * Copyright (C) 2020 Collabora Ltd.
+ *  Author: Guillaume Desmottes <guillaume.desmottes@collabora.com>, Collabora Ltd.
+ *
+ * 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.
+ */
+
+/**
+ * SECTION:element-isacpay
+ * @title: isacpay
+ * @short_description: iSAC RTP Payloader
+ *
+ * Since: 1.20
+ *
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/rtp/gstrtpbuffer.h>
+
+#include "gstrtpisacpay.h"
+#include "gstrtputils.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpisacpay_debug);
+#define GST_CAT_DEFAULT (rtpisacpay_debug)
+
+static GstStaticPadTemplate gst_rtp_isac_pay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/isac, "
+        "rate = (int) { 16000, 32000 }, " "channels = (int) 1")
+    );
+
+static GstStaticPadTemplate gst_rtp_isac_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) { 16000, 32000 }, "
+        "encoding-name = (string) \"ISAC\", "
+        "encoding-params = (string) \"1\"")
+    );
+
+struct _GstRtpIsacPay
+{
+  /*< private > */
+  GstRTPBasePayload parent;
+};
+
+#define gst_rtp_isac_pay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpIsacPay, gst_rtp_isac_pay, GST_TYPE_RTP_BASE_PAYLOAD);
+
+static GstCaps *
+gst_rtp_isac_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;
+      const GValue *v;
+
+      ps = gst_caps_get_structure (otherpadcaps, 0);
+      caps = gst_caps_make_writable (caps);
+      s = gst_caps_get_structure (caps, 0);
+
+      v = gst_structure_get_value (ps, "clock-rate");
+      if (v)
+        gst_structure_set_value (s, "rate", v);
+    }
+    gst_caps_unref (otherpadcaps);
+  }
+
+  if (filter) {
+    GstCaps *tcaps = caps;
+
+    caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST);
+    gst_caps_unref (tcaps);
+  }
+
+  GST_DEBUG_OBJECT (payload, "%" GST_PTR_FORMAT, caps);
+
+  return caps;
+}
+
+static gboolean
+gst_rtp_isac_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
+{
+  GstStructure *s;
+  gint rate;
+
+  GST_DEBUG_OBJECT (payload, "%" GST_PTR_FORMAT, caps);
+
+  s = gst_caps_get_structure (caps, 0);
+  if (!gst_structure_get_int (s, "rate", &rate)) {
+    GST_ERROR_OBJECT (payload, "Missing 'rate' in caps");
+    return FALSE;
+  }
+
+  gst_rtp_base_payload_set_options (payload, "audio", TRUE, "ISAC", rate);
+
+  return gst_rtp_base_payload_set_outcaps (payload, NULL);
+}
+
+static GstFlowReturn
+gst_rtp_isac_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;
+
+  return gst_rtp_base_payload_push (basepayload, outbuf);
+}
+
+static void
+gst_rtp_isac_pay_class_init (GstRtpIsacPayClass * klass)
+{
+  GstElementClass *gstelement_class = (GstElementClass *) klass;
+  GstRTPBasePayloadClass *payload_class = (GstRTPBasePayloadClass *) klass;
+
+  payload_class->get_caps = gst_rtp_isac_pay_getcaps;
+  payload_class->set_caps = gst_rtp_isac_pay_setcaps;
+  payload_class->handle_buffer = gst_rtp_isac_pay_handle_buffer;
+
+  gst_element_class_add_static_pad_template (gstelement_class,
+      &gst_rtp_isac_pay_sink_template);
+  gst_element_class_add_static_pad_template (gstelement_class,
+      &gst_rtp_isac_pay_src_template);
+
+  gst_element_class_set_static_metadata (gstelement_class,
+      "RTP iSAC payloader", "Codec/Payloader/Network/RTP",
+      "Payload-encodes iSAC audio into a RTP packet",
+      "Guillaume Desmottes <guillaume.desmottes@collabora.com>");
+
+  GST_DEBUG_CATEGORY_INIT (rtpisacpay_debug, "rtpisacpay", 0,
+      "iSAC RTP Payloader");
+}
+
+static void
+gst_rtp_isac_pay_init (GstRtpIsacPay * rtpisacpay)
+{
+}
+
+gboolean
+gst_rtp_isac_pay_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpisacpay",
+      GST_RANK_SECONDARY, GST_TYPE_RTP_ISAC_PAY);
+}

diff --git a/gst/rtp/gstrtpisacpay.h b/gst/rtp/gstrtpisacpay.h
new file mode 100644 (file)
index 0000000..764fae4
--- /dev/null
+++ b/gst/rtp/gstrtpisacpay.h
@@ -0,0 +1,33 @@
+/* GStreamer
+ * Copyright (C) 2020 Collabora Ltd.
+ *  Author: Guillaume Desmottes <guillaume.desmottes@collabora.com>, Collabora Ltd.
+ *
+ * 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
+ */
+
+
+#ifndef __GST_RTP_ISAC_PAY_H__
+#define __GST_RTP_ISAC_PAY_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstrtpbasepayload.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTP_ISAC_PAY gst_rtp_isac_pay_get_type ()
+
+G_DECLARE_FINAL_TYPE(GstRtpIsacPay, gst_rtp_isac_pay, GST, RTP_ISAC_PAY, GstRTPBasePayload);
+
+gboolean gst_rtp_isac_pay_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_ISAC_PAY_H__ */

diff --git a/gst/rtp/meson.build b/gst/rtp/meson.build
index d57a195..f0825fa 100644 (file)
--- a/gst/rtp/meson.build
+++ b/gst/rtp/meson.build
@@ -102,6 +102,7 @@ rtp_sources = [
   'rtpstorage.c',
   'rtpstoragestream.c',
   'gstrtpstorage.c',
+  'gstrtpisacpay.c',
 ]
 
 rtp_args = [