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0dbe0e21fe
Multiplying elements named after RFC numbers is confusing, so let's give them meaningful names. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1125>
268 lines
8.1 KiB
C
268 lines
8.1 KiB
C
/* GStreamer
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* Copyright (C) <2018> Havard Graff <havard.graff@gmail.com>
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* Copyright (C) <2020-2021> Guillaume Desmottes <guillaume.desmottes@collabora.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more
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*/
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/**
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* SECTION:element-rtphdrextclientaudiolevel
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* @title: rtphdrextclientaudiolevel
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* @short_description: Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension
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*
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* Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension.
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* The extension should be automatically created by payloader and depayloaders,
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* if their `auto-header-extension` property is enabled, if the extension
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* is part of the RTP caps.
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*
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* ## Example pipeline
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* |[
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* gst-launch-1.0 pulsesrc ! level audio-level-meta=true ! audiconvert !
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* rtpL16pay ! application/x-rtp,
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* extmap-1=(string)\< \"\", urn:ietf:params:rtp-hdrext:ssrc-audio-level,
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* \"vad=on\" \> ! udpsink
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* ]|
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*
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* Since: 1.20
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstrtphdrext-clientaudiolevel.h"
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#include <gst/audio/audio.h>
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#define CLIENT_AUDIO_LEVEL_HDR_EXT_URI GST_RTP_HDREXT_BASE"ssrc-audio-level"
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GST_DEBUG_CATEGORY_STATIC (rtphdrclient_audio_level_debug);
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#define GST_CAT_DEFAULT (rtphdrclient_audio_level_debug)
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#define DEFAULT_VAD TRUE
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enum
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{
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PROP_0,
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PROP_VAD,
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};
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struct _GstRTPHeaderExtensionClientAudioLevel
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{
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GstRTPHeaderExtension parent;
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gboolean vad;
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};
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G_DEFINE_TYPE_WITH_CODE (GstRTPHeaderExtensionClientAudioLevel,
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gst_rtp_header_extension_client_audio_level, GST_TYPE_RTP_HEADER_EXTENSION,
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GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "rtphdrextclientaudiolevel", 0,
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"RTP RFC 6464 Header Extensions"););
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GST_ELEMENT_REGISTER_DEFINE (rtphdrextclientaudiolevel,
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"rtphdrextclientaudiolevel", GST_RANK_MARGINAL,
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GST_TYPE_RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL);
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static void
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gst_rtp_header_extension_client_audio_level_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec)
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{
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GstRTPHeaderExtensionClientAudioLevel *self =
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GST_RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL (object);
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switch (prop_id) {
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case PROP_VAD:
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g_value_set_boolean (value, self->vad);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static GstRTPHeaderExtensionFlags
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gst_rtp_header_extension_client_audio_level_get_supported_flags
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(GstRTPHeaderExtension * ext)
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{
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return GST_RTP_HEADER_EXTENSION_ONE_BYTE | GST_RTP_HEADER_EXTENSION_TWO_BYTE;
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}
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static gsize
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gst_rtp_header_extension_client_audio_level_get_max_size (GstRTPHeaderExtension
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* ext, const GstBuffer * input_meta)
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{
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return 2;
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}
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static void
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set_vad (GstRTPHeaderExtension * ext, gboolean vad)
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{
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GstRTPHeaderExtensionClientAudioLevel *self =
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GST_RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL (ext);
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if (self->vad == vad)
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return;
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GST_DEBUG_OBJECT (ext, "vad: %d", vad);
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self->vad = vad;
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g_object_notify (G_OBJECT (self), "vad");
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}
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static gboolean
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gst_rtp_header_extension_client_audio_level_set_attributes
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(GstRTPHeaderExtension * ext, GstRTPHeaderExtensionDirection direction,
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const gchar * attributes)
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{
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if (g_str_equal (attributes, "vad=on") || g_str_equal (attributes, "")) {
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set_vad (ext, TRUE);
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} else if (g_str_equal (attributes, "vad=off")) {
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set_vad (ext, FALSE);
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} else {
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GST_WARNING_OBJECT (ext, "Invalid attribute: %s", attributes);
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return FALSE;
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}
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return TRUE;
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}
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static gboolean
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gst_rtp_header_extension_client_audio_level_set_caps_from_attributes
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(GstRTPHeaderExtension * ext, GstCaps * caps)
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{
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GstRTPHeaderExtensionClientAudioLevel *self =
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GST_RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL (ext);
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const gchar *vad;
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if (self->vad)
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vad = "vad=on";
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else
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vad = "vad=off";
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return gst_rtp_header_extension_set_caps_from_attributes_helper (ext, caps,
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vad);
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}
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static gssize
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gst_rtp_header_extension_client_audio_level_write (GstRTPHeaderExtension * ext,
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const GstBuffer * input_meta, GstRTPHeaderExtensionFlags write_flags,
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GstBuffer * output, guint8 * data, gsize size)
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{
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GstAudioLevelMeta *meta;
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guint level;
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g_return_val_if_fail (size >=
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gst_rtp_header_extension_client_audio_level_get_max_size (ext, NULL), -1);
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g_return_val_if_fail (write_flags &
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gst_rtp_header_extension_client_audio_level_get_supported_flags (ext),
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-1);
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meta = gst_buffer_get_audio_level_meta ((GstBuffer *) input_meta);
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if (!meta) {
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GST_LOG_OBJECT (ext, "no meta");
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return 0;
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}
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level = meta->level;
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if (level > 127) {
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GST_LOG_OBJECT (ext, "level from meta is higher than 127: %d, cropping",
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meta->level);
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level = 127;
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}
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GST_LOG_OBJECT (ext, "writing ext (level: %d voice: %d)", meta->level,
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meta->voice_activity);
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/* Both one & two byte use the same format, the second byte being padding */
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data[0] = (meta->level & 0x7F) | (meta->voice_activity << 7);
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if (write_flags & GST_RTP_HEADER_EXTENSION_ONE_BYTE) {
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return 1;
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}
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data[1] = 0;
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return 2;
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}
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static gboolean
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gst_rtp_header_extension_client_audio_level_read (GstRTPHeaderExtension * ext,
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GstRTPHeaderExtensionFlags read_flags, const guint8 * data, gsize size,
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GstBuffer * buffer)
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{
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guint8 level;
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gboolean voice_activity;
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g_return_val_if_fail (read_flags &
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gst_rtp_header_extension_client_audio_level_get_supported_flags (ext),
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-1);
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/* Both one & two byte use the same format, the second byte being padding */
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level = data[0] & 0x7F;
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voice_activity = (data[0] & 0x80) >> 7;
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GST_LOG_OBJECT (ext, "reading ext (level: %d voice: %d)", level,
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voice_activity);
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gst_buffer_add_audio_level_meta (buffer, level, voice_activity);
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return TRUE;
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}
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static void
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gst_rtp_header_extension_client_audio_level_class_init
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(GstRTPHeaderExtensionClientAudioLevelClass * klass)
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{
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GstRTPHeaderExtensionClass *rtp_hdr_class;
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GstElementClass *gstelement_class;
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GObjectClass *gobject_class;
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rtp_hdr_class = GST_RTP_HEADER_EXTENSION_CLASS (klass);
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gobject_class = (GObjectClass *) klass;
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gstelement_class = GST_ELEMENT_CLASS (klass);
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gobject_class->get_property =
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gst_rtp_header_extension_client_audio_level_get_property;
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/**
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* rtphdrextclientaudiolevel:vad:
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*
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* If the vad extension attribute is enabled or not, default to %FALSE.
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*
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* Since: 1.20
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*/
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g_object_class_install_property (gobject_class, PROP_VAD,
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g_param_spec_boolean ("vad", "vad",
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"If the vad extension attribute is enabled or not",
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DEFAULT_VAD, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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rtp_hdr_class->get_supported_flags =
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gst_rtp_header_extension_client_audio_level_get_supported_flags;
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rtp_hdr_class->get_max_size =
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gst_rtp_header_extension_client_audio_level_get_max_size;
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rtp_hdr_class->set_attributes =
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gst_rtp_header_extension_client_audio_level_set_attributes;
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rtp_hdr_class->set_caps_from_attributes =
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gst_rtp_header_extension_client_audio_level_set_caps_from_attributes;
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rtp_hdr_class->write = gst_rtp_header_extension_client_audio_level_write;
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rtp_hdr_class->read = gst_rtp_header_extension_client_audio_level_read;
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gst_element_class_set_static_metadata (gstelement_class,
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"Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension",
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GST_RTP_HDREXT_ELEMENT_CLASS,
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"Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension",
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"Guillaume Desmottes <guillaume.desmottes@collabora.com>");
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gst_rtp_header_extension_class_set_uri (rtp_hdr_class,
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CLIENT_AUDIO_LEVEL_HDR_EXT_URI);
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}
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static void
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gst_rtp_header_extension_client_audio_level_init
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(GstRTPHeaderExtensionClientAudioLevel * self)
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{
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GST_DEBUG_OBJECT (self, "creating element");
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self->vad = DEFAULT_VAD;
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}
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