gstreamer/subprojects/gst-plugins-good/gst/rtpmanager/gstrtphdrext-clientaudiolevel.c

269 lines
8.1 KiB
C
Raw Normal View History

/* GStreamer
* Copyright (C) <2018> Havard Graff <havard.graff@gmail.com>
* Copyright (C) <2020-2021> Guillaume Desmottes <guillaume.desmottes@collabora.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
*/
/**
* SECTION:element-rtphdrextclientaudiolevel
* @title: rtphdrextclientaudiolevel
* @short_description: Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension
*
* Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension.
* The extension should be automatically created by payloader and depayloaders,
* if their `auto-header-extension` property is enabled, if the extension
* is part of the RTP caps.
*
* ## Example pipeline
* |[
* gst-launch-1.0 pulsesrc ! level audio-level-meta=true ! audiconvert !
* rtpL16pay ! application/x-rtp,
* extmap-1=(string)\< \"\", urn:ietf:params:rtp-hdrext:ssrc-audio-level,
* \"vad=on\" \> ! udpsink
* ]|
*
* Since: 1.20
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstrtphdrext-clientaudiolevel.h"
#include <gst/audio/audio.h>
#define CLIENT_AUDIO_LEVEL_HDR_EXT_URI GST_RTP_HDREXT_BASE"ssrc-audio-level"
GST_DEBUG_CATEGORY_STATIC (rtphdrclient_audio_level_debug);
#define GST_CAT_DEFAULT (rtphdrclient_audio_level_debug)
#define DEFAULT_VAD TRUE
enum
{
PROP_0,
PROP_VAD,
};
struct _GstRTPHeaderExtensionClientAudioLevel
{
GstRTPHeaderExtension parent;
gboolean vad;
};
G_DEFINE_TYPE_WITH_CODE (GstRTPHeaderExtensionClientAudioLevel,
gst_rtp_header_extension_client_audio_level, GST_TYPE_RTP_HEADER_EXTENSION,
GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "rtphdrextclientaudiolevel", 0,
"RTP RFC 6464 Header Extensions"););
GST_ELEMENT_REGISTER_DEFINE (rtphdrextclientaudiolevel,
"rtphdrextclientaudiolevel", GST_RANK_MARGINAL,
GST_TYPE_RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL);
static void
gst_rtp_header_extension_client_audio_level_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstRTPHeaderExtensionClientAudioLevel *self =
GST_RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL (object);
switch (prop_id) {
case PROP_VAD:
g_value_set_boolean (value, self->vad);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstRTPHeaderExtensionFlags
gst_rtp_header_extension_client_audio_level_get_supported_flags
(GstRTPHeaderExtension * ext)
{
return GST_RTP_HEADER_EXTENSION_ONE_BYTE | GST_RTP_HEADER_EXTENSION_TWO_BYTE;
}
static gsize
gst_rtp_header_extension_client_audio_level_get_max_size (GstRTPHeaderExtension
* ext, const GstBuffer * input_meta)
{
return 2;
}
static void
set_vad (GstRTPHeaderExtension * ext, gboolean vad)
{
GstRTPHeaderExtensionClientAudioLevel *self =
GST_RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL (ext);
if (self->vad == vad)
return;
GST_DEBUG_OBJECT (ext, "vad: %d", vad);
self->vad = vad;
g_object_notify (G_OBJECT (self), "vad");
}
static gboolean
gst_rtp_header_extension_client_audio_level_set_attributes
(GstRTPHeaderExtension * ext, GstRTPHeaderExtensionDirection direction,
const gchar * attributes)
{
if (g_str_equal (attributes, "vad=on") || g_str_equal (attributes, "")) {
set_vad (ext, TRUE);
} else if (g_str_equal (attributes, "vad=off")) {
set_vad (ext, FALSE);
} else {
GST_WARNING_OBJECT (ext, "Invalid attribute: %s", attributes);
return FALSE;
}
return TRUE;
}
static gboolean
gst_rtp_header_extension_client_audio_level_set_caps_from_attributes
(GstRTPHeaderExtension * ext, GstCaps * caps)
{
GstRTPHeaderExtensionClientAudioLevel *self =
GST_RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL (ext);
const gchar *vad;
if (self->vad)
vad = "vad=on";
else
vad = "vad=off";
return gst_rtp_header_extension_set_caps_from_attributes_helper (ext, caps,
vad);
}
static gssize
gst_rtp_header_extension_client_audio_level_write (GstRTPHeaderExtension * ext,
const GstBuffer * input_meta, GstRTPHeaderExtensionFlags write_flags,
GstBuffer * output, guint8 * data, gsize size)
{
GstAudioLevelMeta *meta;
guint level;
g_return_val_if_fail (size >=
gst_rtp_header_extension_client_audio_level_get_max_size (ext, NULL), -1);
g_return_val_if_fail (write_flags &
gst_rtp_header_extension_client_audio_level_get_supported_flags (ext),
-1);
meta = gst_buffer_get_audio_level_meta ((GstBuffer *) input_meta);
if (!meta) {
GST_LOG_OBJECT (ext, "no meta");
return 0;
}
level = meta->level;
if (level > 127) {
GST_LOG_OBJECT (ext, "level from meta is higher than 127: %d, cropping",
meta->level);
level = 127;
}
GST_LOG_OBJECT (ext, "writing ext (level: %d voice: %d)", meta->level,
meta->voice_activity);
/* Both one & two byte use the same format, the second byte being padding */
data[0] = (meta->level & 0x7F) | (meta->voice_activity << 7);
if (write_flags & GST_RTP_HEADER_EXTENSION_ONE_BYTE) {
return 1;
}
data[1] = 0;
return 2;
}
static gboolean
gst_rtp_header_extension_client_audio_level_read (GstRTPHeaderExtension * ext,
GstRTPHeaderExtensionFlags read_flags, const guint8 * data, gsize size,
GstBuffer * buffer)
{
guint8 level;
gboolean voice_activity;
g_return_val_if_fail (read_flags &
gst_rtp_header_extension_client_audio_level_get_supported_flags (ext),
-1);
/* Both one & two byte use the same format, the second byte being padding */
level = data[0] & 0x7F;
voice_activity = (data[0] & 0x80) >> 7;
GST_LOG_OBJECT (ext, "reading ext (level: %d voice: %d)", level,
voice_activity);
gst_buffer_add_audio_level_meta (buffer, level, voice_activity);
return TRUE;
}
static void
gst_rtp_header_extension_client_audio_level_class_init
(GstRTPHeaderExtensionClientAudioLevelClass * klass)
{
GstRTPHeaderExtensionClass *rtp_hdr_class;
GstElementClass *gstelement_class;
GObjectClass *gobject_class;
rtp_hdr_class = GST_RTP_HEADER_EXTENSION_CLASS (klass);
gobject_class = (GObjectClass *) klass;
gstelement_class = GST_ELEMENT_CLASS (klass);
gobject_class->get_property =
gst_rtp_header_extension_client_audio_level_get_property;
/**
* rtphdrextclientaudiolevel:vad:
*
* If the vad extension attribute is enabled or not, default to %FALSE.
*
* Since: 1.20
*/
g_object_class_install_property (gobject_class, PROP_VAD,
g_param_spec_boolean ("vad", "vad",
"If the vad extension attribute is enabled or not",
DEFAULT_VAD, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
rtp_hdr_class->get_supported_flags =
gst_rtp_header_extension_client_audio_level_get_supported_flags;
rtp_hdr_class->get_max_size =
gst_rtp_header_extension_client_audio_level_get_max_size;
rtp_hdr_class->set_attributes =
gst_rtp_header_extension_client_audio_level_set_attributes;
rtp_hdr_class->set_caps_from_attributes =
gst_rtp_header_extension_client_audio_level_set_caps_from_attributes;
rtp_hdr_class->write = gst_rtp_header_extension_client_audio_level_write;
rtp_hdr_class->read = gst_rtp_header_extension_client_audio_level_read;
gst_element_class_set_static_metadata (gstelement_class,
"Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension",
GST_RTP_HDREXT_ELEMENT_CLASS,
"Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension",
"Guillaume Desmottes <guillaume.desmottes@collabora.com>");
gst_rtp_header_extension_class_set_uri (rtp_hdr_class,
CLIENT_AUDIO_LEVEL_HDR_EXT_URI);
}
static void
gst_rtp_header_extension_client_audio_level_init
(GstRTPHeaderExtensionClientAudioLevel * self)
{
GST_DEBUG_OBJECT (self, "creating element");
self->vad = DEFAULT_VAD;
}