gstreamer/subprojects/gst-plugins-good/gst/rtpmanager/gstrtphdrext-clientaudiolevel.c
Sanchayan Maity 00bbac6541 rtphdrext-clientaudiolevel: Fix level value being written by the extension
When level value is greater than 127, it was being clamped but this clamped
value was not the one being actually used. For level values greater than 127
this resulted in an incorrect value being used. As an example, a level value
of 187, after and'ed with 0x7F, it would result in 0x3B being reported as the
level value.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5893>
2024-01-07 16:00:18 +05:30

268 lines
8.1 KiB
C

/* 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)", level,
meta->voice_activity);
/* Both one & two byte use the same format, the second byte being padding */
data[0] = (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;
}