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opus: remove Opus RTP elements, they have moved to -good

Browse source 
https://bugzilla.gnome.org/show_bug.cgi?id=756282
This commit is contained in:
Tim-Philipp Müller 2016-02-17 15:20:47 +00:00
parent 35e00becfe
commit a50e4bcadf
10 changed files with 6 additions and 658 deletions
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2
docs/plugins/gst-plugins-bad-plugins-docs.sgml
View file

@ -76,8 +76,6 @@
<xi:include href="xml/element-opusdec.xml" /> <xi:include href="xml/element-opusdec.xml" />
<xi:include href="xml/element-opusenc.xml" /> <xi:include href="xml/element-opusenc.xml" />
<xi:include href="xml/element-opusparse.xml" /> <xi:include href="xml/element-opusparse.xml" />
<xi:include href="xml/element-rtpopuspay.xml" />
<xi:include href="xml/element-rtpopusdepay.xml" />
<xi:include href="xml/element-pcapparse.xml" /> <xi:include href="xml/element-pcapparse.xml" />
<xi:include href="xml/element-pinch.xml" /> <xi:include href="xml/element-pinch.xml" />
<xi:include href="xml/element-pyramidsegment.xml" /> <xi:include href="xml/element-pyramidsegment.xml" />

26
docs/plugins/gst-plugins-bad-plugins-sections.txt
View file

@ -1141,32 +1141,6 @@ GST_IS_OPUS_PARSE
GST_IS_OPUS_PARSE_CLASS GST_IS_OPUS_PARSE_CLASS
</SECTION> </SECTION>
<FILE>element-rtpopusdepay</FILE>
<TITLE>rtpopusdepay</TITLE>
GstRTPOpusDepay
<SUBSECTION Standard>
GstRTPOpusDepayClass
gst_rtp_opus_depay_get_type
GST_TYPE_RTP_OPUS_DEPAY
GST_RTP_OPUS_DEPAY
GST_RTP_OPUS_DEPAY_CLASS
GST_IS_RTP_OPUS_DEPAY
GST_IS_RTP_OPUS_DEPAY_CLASS
</SECTION>
<FILE>element-rtpopuspay</FILE>
<TITLE>rtpopuspay</TITLE>
GstRtpOPUSPay
<SUBSECTION Standard>
GstRtpOPUSPayClass
gst_rtp_opus_pay_get_type
GST_TYPE_RTP_OPUS_PAY
GST_RTP_OPUS_PAY
GST_RTP_OPUS_PAY_CLASS
GST_IS_RTP_OPUS_PAY
GST_IS_RTP_OPUS_PAY_CLASS
</SECTION>
<SECTION> <SECTION>
<FILE>element-pcapparse</FILE> <FILE>element-pcapparse</FILE>
<TITLE>pcapparse</TITLE> <TITLE>pcapparse</TITLE>

3
docs/plugins/gst-plugins-bad-plugins.hierarchy
View file

@ -303,11 +303,8 @@ GObject
GstNetSim GstNetSim
GstPcapParse GstPcapParse
GstPitch GstPitch
GstRTPBaseDepayload
GstRTPOpusDepay
GstRTPBasePayload GstRTPBasePayload
GstRtpAsfPay GstRtpAsfPay
GstRtpOPUSPay
GstRawParse GstRawParse
GstAudioParse GstAudioParse
GstVideoParse GstVideoParse

48
docs/plugins/inspect/plugin-opus.xml
View file

@ -3,10 +3,10 @@
<description>OPUS plugin library</description> <description>OPUS plugin library</description>
<filename>../../ext/opus/.libs/libgstopus.so</filename> <filename>../../ext/opus/.libs/libgstopus.so</filename>
<basename>libgstopus.so</basename> <basename>libgstopus.so</basename>
<version>1.7.2</version> <version>1.7.2.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-bad</source> <source>gst-plugins-bad</source>
<package>GStreamer Bad Plug-ins source release</package> <package>GStreamer Bad Plug-ins git</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>
@ -68,49 +68,7 @@
<name>src</name> <name>src</name>
<direction>source</direction> <direction>source</direction>
<presence>always</presence> <presence>always</presence>
<details>audio/x-opus, framed=(boolean)true</details> <details>audio/x-opus</details>
</caps>
</pads>
</element>
<element>
<name>rtpopusdepay</name>
<longname>RTP Opus packet depayloader</longname>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts Opus audio from RTP packets</description>
<author>Danilo Cesar Lemes de Paula &lt;danilo.cesar@collabora.co.uk&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)48000, encoding-name=(string){ OPUS, X-GST-OPUS-DRAFT-SPITTKA-00 }</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-opus, channel-mapping-family=(int)0</details>
</caps>
</pads>
</element>
<element>
<name>rtpopuspay</name>
<longname>RTP Opus payloader</longname>
<class>Codec/Payloader/Network/RTP</class>
<description>Puts Opus audio in RTP packets</description>
<author>Danilo Cesar Lemes de Paula &lt;danilo.cesar@collabora.co.uk&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-opus, channels=(int)[ 1, 2 ], channel-mapping-family=(int)0</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)48000, encoding-params=(string)2, encoding-name=(string){ OPUS, X-GST-OPUS-DRAFT-SPITTKA-00 }</details>
</caps> </caps>
</pads> </pads>
</element> </element>

6
ext/opus/Makefile.am
View file

@ -1,6 +1,6 @@
plugin_LTLIBRARIES = libgstopus.la plugin_LTLIBRARIES = libgstopus.la
libgstopus_la_SOURCES = gstopus.c gstopusdec.c gstopusenc.c gstopusparse.c gstopusheader.c gstopuscommon.c gstrtpopuspay.c gstrtpopusdepay.c libgstopus_la_SOURCES = gstopus.c gstopusdec.c gstopusenc.c gstopusparse.c gstopusheader.c gstopuscommon.c
libgstopus_la_CFLAGS = \ libgstopus_la_CFLAGS = \
-DGST_USE_UNSTABLE_API \ -DGST_USE_UNSTABLE_API \
$(GST_PLUGINS_BAD_CFLAGS) \ $(GST_PLUGINS_BAD_CFLAGS) \
@ -9,7 +9,7 @@ libgstopus_la_CFLAGS = \
$(OPUS_CFLAGS) $(OPUS_CFLAGS)
libgstopus_la_LIBADD = \ libgstopus_la_LIBADD = \
$(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) \ $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) \
-lgsttag-$(GST_API_VERSION) -lgstrtp-$(GST_API_VERSION) \ -lgsttag-$(GST_API_VERSION) \
-lgstpbutils-$(GST_API_VERSION) \ -lgstpbutils-$(GST_API_VERSION) \
$(GST_BASE_LIBS) \ $(GST_BASE_LIBS) \
$(GST_LIBS) \ $(GST_LIBS) \
@ -17,4 +17,4 @@ libgstopus_la_LIBADD = \
libgstopus_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(LIBM) libgstopus_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(LIBM)
libgstopus_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS) libgstopus_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstopusenc.h gstopusdec.h gstopusparse.h gstopusheader.h gstopuscommon.h gstrtpopuspay.h gstrtpopusdepay.h noinst_HEADERS = gstopusenc.h gstopusdec.h gstopusparse.h gstopusheader.h gstopuscommon.h

11
ext/opus/gstopus.c
View file

@ -25,9 +25,6 @@
#include "gstopusenc.h" #include "gstopusenc.h"
#include "gstopusparse.h" #include "gstopusparse.h"
#include "gstrtpopuspay.h"
#include "gstrtpopusdepay.h"
#include <gst/tag/tag.h> #include <gst/tag/tag.h>
static gboolean static gboolean
@ -46,14 +43,6 @@ plugin_init (GstPlugin * plugin)
GST_TYPE_OPUS_PARSE)) GST_TYPE_OPUS_PARSE))
return FALSE; return FALSE;
if (!gst_element_register (plugin, "rtpopusdepay", GST_RANK_SECONDARY,
GST_TYPE_RTP_OPUS_DEPAY))
return FALSE;
if (!gst_element_register (plugin, "rtpopuspay", GST_RANK_SECONDARY,
GST_TYPE_RTP_OPUS_PAY))
return FALSE;
gst_tag_register_musicbrainz_tags (); gst_tag_register_musicbrainz_tags ();
return TRUE; return TRUE;

175
ext/opus/gstrtpopusdepay.c
View file

@ -1,175 +0,0 @@
/*
* 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 "gstrtpopusdepay.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\" }")
);
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")
);
static GstBuffer *gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload,
GstBuffer * buf);
static gboolean gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
G_DEFINE_TYPE (GstRTPOpusDepay, gst_rtp_opus_depay,
GST_TYPE_RTP_BASE_DEPAYLOAD);
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_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_opus_depay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&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 = 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_stereo, *sprop_maxcapturerate;
srccaps =
gst_caps_new_simple ("audio/x-opus", "channel-mapping-family", G_TYPE_INT,
0, NULL);
s = gst_caps_get_structure (caps, 0);
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);
}
if ((sprop_maxcapturerate =
gst_structure_get_string (s, "sprop-maxcapturerate"))) {
gulong rate;
gchar *tailptr;
rate = strtoul (sprop_maxcapturerate, &tailptr, 10);
if (rate > INT_MAX || *tailptr != '\0') {
GST_WARNING_OBJECT (depayload,
"Failed to parse sprop-maxcapturerate value '%s'",
sprop_maxcapturerate);
} else {
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;
}
static gboolean
foreach_metadata (GstBuffer * inbuf, GstMeta ** meta, gpointer user_data)
{
GstRTPOpusDepay *depay = user_data;
const GstMetaInfo *info = (*meta)->info;
const gchar *const *tags = gst_meta_api_type_get_tags (info->api);
if (!tags || (g_strv_length ((gchar **) tags) == 1
&& gst_meta_api_type_has_tag (info->api,
g_quark_from_string (GST_META_TAG_AUDIO_STR)))) {
GST_DEBUG_OBJECT (depay, "keeping metadata %s", g_type_name (info->api));
} else {
GST_DEBUG_OBJECT (depay, "dropping metadata %s", g_type_name (info->api));
*meta = NULL;
}
return TRUE;
}
static GstBuffer *
gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstBuffer *outbuf;
GstRTPBuffer rtpbuf = { NULL, };
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuf);
outbuf = gst_rtp_buffer_get_payload_buffer (&rtpbuf);
gst_rtp_buffer_unmap (&rtpbuf);
outbuf = gst_buffer_make_writable (outbuf);
/* Filter away all metas that are not sensible to copy */
gst_buffer_foreach_meta (outbuf, foreach_metadata, depayload);
return outbuf;
}

57
ext/opus/gstrtpopusdepay.h
View file

@ -1,57 +0,0 @@
/*
* Opus Depayloader Gst Element
*
* @author: Danilo Cesar Lemes de Paula <danilo.eu@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.
*/
#ifndef __GST_RTP_OPUS_DEPAY_H__
#define __GST_RTP_OPUS_DEPAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstrtpbasedepayload.h>
G_BEGIN_DECLS typedef struct _GstRTPOpusDepay GstRTPOpusDepay;
typedef struct _GstRTPOpusDepayClass GstRTPOpusDepayClass;
#define GST_TYPE_RTP_OPUS_DEPAY \
(gst_rtp_opus_depay_get_type())
#define GST_RTP_OPUS_DEPAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_OPUS_DEPAY,GstRTPOpusDepay))
#define GST_RTP_OPUS_DEPAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_OPUS_DEPAY,GstRTPOpusDepayClass))
#define GST_IS_RTP_OPUS_DEPAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_OPUS_DEPAY))
#define GST_IS_RTP_OPUS_DEPAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_OPUS_DEPAY))
struct _GstRTPOpusDepay
{
GstRTPBaseDepayload depayload;
};
struct _GstRTPOpusDepayClass
{
GstRTPBaseDepayloadClass parent_class;
};
GType gst_rtp_opus_depay_get_type (void);
G_END_DECLS
#endif /* __GST_RTP_OPUS_DEPAY_H__ */

278
ext/opus/gstrtpopuspay.c
View file

@ -1,278 +0,0 @@
/*
* 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"
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, channels = (int) [1, 2], 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_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_opus_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&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;
char *encoding_name;
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) {
src_caps = gst_caps_make_writable (src_caps);
src_caps = gst_caps_truncate (src_caps);
s = gst_caps_get_structure (src_caps, 0);
gst_structure_fixate_field_string (s, "encoding-name", "OPUS");
encoding_name = g_strdup (gst_structure_get_string (s, "encoding-name"));
gst_caps_unref (src_caps);
} else {
encoding_name = g_strdup ("X-GST-OPUS-DRAFT-SPITTKA-00");
}
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);
g_free (encoding_name);
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;
}
typedef struct
{
GstRtpOPUSPay *pay;
GstBuffer *outbuf;
} CopyMetaData;
static gboolean
foreach_metadata (GstBuffer * inbuf, GstMeta ** meta, gpointer user_data)
{
CopyMetaData *data = user_data;
GstRtpOPUSPay *pay = data->pay;
GstBuffer *outbuf = data->outbuf;
const GstMetaInfo *info = (*meta)->info;
const gchar *const *tags = gst_meta_api_type_get_tags (info->api);
if (!tags || (g_strv_length ((gchar **) tags) == 1
&& gst_meta_api_type_has_tag (info->api,
g_quark_from_string (GST_META_TAG_AUDIO_STR)))) {
GstMetaTransformCopy copy_data = { FALSE, 0, -1 };
GST_DEBUG_OBJECT (pay, "copy metadata %s", g_type_name (info->api));
/* simply copy then */
info->transform_func (outbuf, *meta, inbuf,
_gst_meta_transform_copy, &copy_data);
} else {
GST_DEBUG_OBJECT (pay, "not copying metadata %s", g_type_name (info->api));
}
return TRUE;
}
static GstFlowReturn
gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstBuffer *outbuf;
GstClockTime pts, dts, duration;
CopyMetaData data;
pts = GST_BUFFER_PTS (buffer);
dts = GST_BUFFER_DTS (buffer);
duration = GST_BUFFER_DURATION (buffer);
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
data.pay = GST_RTP_OPUS_PAY (basepayload);
data.outbuf = outbuf;
gst_buffer_foreach_meta (buffer, foreach_metadata, &data);
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;
}

58
ext/opus/gstrtpopuspay.h
View file

@ -1,58 +0,0 @@
/*
* Opus Payloader Gst Element
*
* @author: Danilo Cesar Lemes de Paula <danilo.eu@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.
*/
#ifndef __GST_RTP_OPUS_PAY_H__
#define __GST_RTP_OPUS_PAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstrtpbasepayload.h>
G_BEGIN_DECLS
#define GST_TYPE_RTP_OPUS_PAY \
(gst_rtp_opus_pay_get_type())
#define GST_RTP_OPUS_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_OPUS_PAY,GstRtpOPUSPay))
#define GST_RTP_OPUS_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_OPUS_PAY,GstRtpOPUSPayClass))
#define GST_IS_RTP_OPUS_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_OPUS_PAY))
#define GST_IS_RTP_OPUS_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_OPUS_PAY))
typedef struct _GstRtpOPUSPay GstRtpOPUSPay;
typedef struct _GstRtpOPUSPayClass GstRtpOPUSPayClass;
struct _GstRtpOPUSPay
{
GstRTPBasePayload payload;
};
struct _GstRtpOPUSPayClass
{
GstRTPBasePayloadClass parent_class;
};
GType gst_rtp_opus_pay_get_type (void);
G_END_DECLS
#endif /* __GST_RTP_OPUS_PAY_H__ */