diff --git a/subprojects/gst-docs/symbols/symbol_index.json b/subprojects/gst-docs/symbols/symbol_index.json index 89920f0804..c6ef74ca68 100644 --- a/subprojects/gst-docs/symbols/symbol_index.json +++ b/subprojects/gst-docs/symbols/symbol_index.json @@ -63063,8 +63063,8 @@ "rtph265pay", "rtph265pay:aggregate-mode", "rtph265pay:config-interval", - "rtphdrextrfc6464", - "rtphdrextrfc6464:vad", + "rtphdrextclientaudiolevel", + "rtphdrextclientaudiolevel:vad", "rtphdrexttwcc", "rtphdrexttwcc:n-streams", "rtpilbcdepay", diff --git a/subprojects/gst-plugins-good/docs/gst_plugins_cache.json b/subprojects/gst-plugins-good/docs/gst_plugins_cache.json index 2e404619ba..84a396eb46 100644 --- a/subprojects/gst-plugins-good/docs/gst_plugins_cache.json +++ b/subprojects/gst-plugins-good/docs/gst_plugins_cache.json @@ -17162,12 +17162,12 @@ }, "rank": "none" }, - "rtphdrextrfc6464": { + "rtphdrextclientaudiolevel": { "RTP-Header-Extension-URI": "urn:ietf:params:rtp-hdrext:ssrc-audio-level", "author": "Guillaume Desmottes ", "description": "Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension", "hierarchy": [ - "GstRTPHeaderExtensionRfc6464", + "GstRTPHeaderExtensionClientAudioLevel", "GstRTPHeaderExtension", "GstElement", "GstObject", diff --git a/subprojects/gst-plugins-good/gst/rtpmanager/gstrtphdrext-rfc6464.c b/subprojects/gst-plugins-good/gst/rtpmanager/gstrtphdrext-clientaudiolevel.c similarity index 63% rename from subprojects/gst-plugins-good/gst/rtpmanager/gstrtphdrext-rfc6464.c rename to subprojects/gst-plugins-good/gst/rtpmanager/gstrtphdrext-clientaudiolevel.c index d93fcaffd1..4012e18bc2 100644 --- a/subprojects/gst-plugins-good/gst/rtpmanager/gstrtphdrext-rfc6464.c +++ b/subprojects/gst-plugins-good/gst/rtpmanager/gstrtphdrext-clientaudiolevel.c @@ -14,8 +14,8 @@ */ /** - * SECTION:element-rtphdrextrfc6464 - * @title: rtphdrextrfc6464 + * 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. @@ -38,14 +38,14 @@ #include "config.h" #endif -#include "gstrtphdrext-rfc6464.h" +#include "gstrtphdrext-clientaudiolevel.h" #include -#define RFC6464_HDR_EXT_URI GST_RTP_HDREXT_BASE"ssrc-audio-level" +#define CLIENT_AUDIO_LEVEL_HDR_EXT_URI GST_RTP_HDREXT_BASE"ssrc-audio-level" -GST_DEBUG_CATEGORY_STATIC (rtphdrrfc6464_twcc_debug); -#define GST_CAT_DEFAULT (rtphdrrfc6464_twcc_debug) +GST_DEBUG_CATEGORY_STATIC (rtphdrclient_audio_level_debug); +#define GST_CAT_DEFAULT (rtphdrclient_audio_level_debug) #define DEFAULT_VAD TRUE @@ -55,26 +55,27 @@ enum PROP_VAD, }; -struct _GstRTPHeaderExtensionRfc6464 +struct _GstRTPHeaderExtensionClientAudioLevel { GstRTPHeaderExtension parent; gboolean vad; }; -G_DEFINE_TYPE_WITH_CODE (GstRTPHeaderExtensionRfc6464, - gst_rtp_header_extension_rfc6464, GST_TYPE_RTP_HEADER_EXTENSION, - GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "rtphdrextrfc6464", 0, +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 (rtphdrextrfc6464, "rtphdrextrfc6464", - GST_RANK_MARGINAL, GST_TYPE_RTP_HEADER_EXTENSION_RFC6464); +GST_ELEMENT_REGISTER_DEFINE (rtphdrextclientaudiolevel, + "rtphdrextclientaudiolevel", GST_RANK_MARGINAL, + GST_TYPE_RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL); static void -gst_rtp_header_extension_rfc6464_get_property (GObject * object, +gst_rtp_header_extension_client_audio_level_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { - GstRTPHeaderExtensionRfc6464 *self = - GST_RTP_HEADER_EXTENSION_RFC6464 (object); + GstRTPHeaderExtensionClientAudioLevel *self = + GST_RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL (object); switch (prop_id) { case PROP_VAD: @@ -87,15 +88,15 @@ gst_rtp_header_extension_rfc6464_get_property (GObject * object, } static GstRTPHeaderExtensionFlags -gst_rtp_header_extension_rfc6464_get_supported_flags (GstRTPHeaderExtension * - ext) + 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_rfc6464_get_max_size (GstRTPHeaderExtension * ext, - const GstBuffer * input_meta) +gst_rtp_header_extension_client_audio_level_get_max_size (GstRTPHeaderExtension + * ext, const GstBuffer * input_meta) { return 2; } @@ -103,7 +104,8 @@ gst_rtp_header_extension_rfc6464_get_max_size (GstRTPHeaderExtension * ext, static void set_vad (GstRTPHeaderExtension * ext, gboolean vad) { - GstRTPHeaderExtensionRfc6464 *self = GST_RTP_HEADER_EXTENSION_RFC6464 (ext); + GstRTPHeaderExtensionClientAudioLevel *self = + GST_RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL (ext); if (self->vad == vad) return; @@ -114,8 +116,9 @@ set_vad (GstRTPHeaderExtension * ext, gboolean vad) } static gboolean -gst_rtp_header_extension_rfc6464_set_attributes (GstRTPHeaderExtension * ext, - GstRTPHeaderExtensionDirection direction, const gchar * attributes) + 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); @@ -130,10 +133,11 @@ gst_rtp_header_extension_rfc6464_set_attributes (GstRTPHeaderExtension * ext, } static gboolean -gst_rtp_header_extension_rfc6464_set_caps_from_attributes (GstRTPHeaderExtension - * ext, GstCaps * caps) + gst_rtp_header_extension_client_audio_level_set_caps_from_attributes + (GstRTPHeaderExtension * ext, GstCaps * caps) { - GstRTPHeaderExtensionRfc6464 *self = GST_RTP_HEADER_EXTENSION_RFC6464 (ext); + GstRTPHeaderExtensionClientAudioLevel *self = + GST_RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL (ext); const gchar *vad; if (self->vad) @@ -146,7 +150,7 @@ gst_rtp_header_extension_rfc6464_set_caps_from_attributes (GstRTPHeaderExtension } static gssize -gst_rtp_header_extension_rfc6464_write (GstRTPHeaderExtension * ext, +gst_rtp_header_extension_client_audio_level_write (GstRTPHeaderExtension * ext, const GstBuffer * input_meta, GstRTPHeaderExtensionFlags write_flags, GstBuffer * output, guint8 * data, gsize size) { @@ -154,9 +158,10 @@ gst_rtp_header_extension_rfc6464_write (GstRTPHeaderExtension * ext, guint level; g_return_val_if_fail (size >= - gst_rtp_header_extension_rfc6464_get_max_size (ext, NULL), -1); + gst_rtp_header_extension_client_audio_level_get_max_size (ext, NULL), -1); g_return_val_if_fail (write_flags & - gst_rtp_header_extension_rfc6464_get_supported_flags (ext), -1); + gst_rtp_header_extension_client_audio_level_get_supported_flags (ext), + -1); meta = gst_buffer_get_audio_level_meta ((GstBuffer *) input_meta); if (!meta) { @@ -184,7 +189,7 @@ gst_rtp_header_extension_rfc6464_write (GstRTPHeaderExtension * ext, } static gboolean -gst_rtp_header_extension_rfc6464_read (GstRTPHeaderExtension * ext, +gst_rtp_header_extension_client_audio_level_read (GstRTPHeaderExtension * ext, GstRTPHeaderExtensionFlags read_flags, const guint8 * data, gsize size, GstBuffer * buffer) { @@ -192,7 +197,8 @@ gst_rtp_header_extension_rfc6464_read (GstRTPHeaderExtension * ext, gboolean voice_activity; g_return_val_if_fail (read_flags & - gst_rtp_header_extension_rfc6464_get_supported_flags (ext), -1); + 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; @@ -207,8 +213,8 @@ gst_rtp_header_extension_rfc6464_read (GstRTPHeaderExtension * ext, } static void -gst_rtp_header_extension_rfc6464_class_init (GstRTPHeaderExtensionRfc6464Class * - klass) + gst_rtp_header_extension_client_audio_level_class_init + (GstRTPHeaderExtensionClientAudioLevelClass * klass) { GstRTPHeaderExtensionClass *rtp_hdr_class; GstElementClass *gstelement_class; @@ -218,10 +224,11 @@ gst_rtp_header_extension_rfc6464_class_init (GstRTPHeaderExtensionRfc6464Class * gobject_class = (GObjectClass *) klass; gstelement_class = GST_ELEMENT_CLASS (klass); - gobject_class->get_property = gst_rtp_header_extension_rfc6464_get_property; + gobject_class->get_property = + gst_rtp_header_extension_client_audio_level_get_property; /** - * rtphdrextrfc6464:vad: + * rtphdrextclientaudiolevel:vad: * * If the vad extension attribute is enabled or not, default to %FALSE. * @@ -233,25 +240,28 @@ gst_rtp_header_extension_rfc6464_class_init (GstRTPHeaderExtensionRfc6464Class * DEFAULT_VAD, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); rtp_hdr_class->get_supported_flags = - gst_rtp_header_extension_rfc6464_get_supported_flags; - rtp_hdr_class->get_max_size = gst_rtp_header_extension_rfc6464_get_max_size; + 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_rfc6464_set_attributes; + gst_rtp_header_extension_client_audio_level_set_attributes; rtp_hdr_class->set_caps_from_attributes = - gst_rtp_header_extension_rfc6464_set_caps_from_attributes; - rtp_hdr_class->write = gst_rtp_header_extension_rfc6464_write; - rtp_hdr_class->read = gst_rtp_header_extension_rfc6464_read; + 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 "); - gst_rtp_header_extension_class_set_uri (rtp_hdr_class, RFC6464_HDR_EXT_URI); + gst_rtp_header_extension_class_set_uri (rtp_hdr_class, + CLIENT_AUDIO_LEVEL_HDR_EXT_URI); } static void -gst_rtp_header_extension_rfc6464_init (GstRTPHeaderExtensionRfc6464 * self) + gst_rtp_header_extension_client_audio_level_init + (GstRTPHeaderExtensionClientAudioLevel * self) { GST_DEBUG_OBJECT (self, "creating element"); self->vad = DEFAULT_VAD; diff --git a/subprojects/gst-plugins-good/gst/rtpmanager/gstrtphdrext-rfc6464.h b/subprojects/gst-plugins-good/gst/rtpmanager/gstrtphdrext-clientaudiolevel.h similarity index 60% rename from subprojects/gst-plugins-good/gst/rtpmanager/gstrtphdrext-rfc6464.h rename to subprojects/gst-plugins-good/gst/rtpmanager/gstrtphdrext-clientaudiolevel.h index f2439a952d..3ca688850c 100644 --- a/subprojects/gst-plugins-good/gst/rtpmanager/gstrtphdrext-rfc6464.h +++ b/subprojects/gst-plugins-good/gst/rtpmanager/gstrtphdrext-clientaudiolevel.h @@ -13,20 +13,20 @@ * Library General Public License for more */ -#ifndef __GST_RTPHDREXT_RFC6464_H__ -#define __GST_RTPHDREXT_RFC6464_H__ +#ifndef __GST_RTPHDREXT_CLIENT_AUDIO_LEVEL_H__ +#define __GST_RTPHDREXT_CLIENT_AUDIO_LEVEL_H__ #include #include G_BEGIN_DECLS -#define GST_TYPE_RTP_HEADER_EXTENSION_RFC6464 (gst_rtp_header_extension_rfc6464_get_type()) +#define GST_TYPE_RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL (gst_rtp_header_extension_client_audio_level_get_type()) -G_DECLARE_FINAL_TYPE (GstRTPHeaderExtensionRfc6464, gst_rtp_header_extension_rfc6464, GST, RTP_HEADER_EXTENSION_RFC6464, GstRTPHeaderExtension) +G_DECLARE_FINAL_TYPE (GstRTPHeaderExtensionClientAudioLevel, gst_rtp_header_extension_client_audio_level, GST, RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL, GstRTPHeaderExtension) -GST_ELEMENT_REGISTER_DECLARE (rtphdrextrfc6464); +GST_ELEMENT_REGISTER_DECLARE (rtphdrextclientaudiolevel); G_END_DECLS -#endif /* __GST_RTPHDREXT_RFC6464_H__ */ +#endif /* __GST_RTPHDREXT_CLIENT_AUDIO_LEVEL_H__ */ diff --git a/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpmanager.c b/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpmanager.c index c3fc29b30b..1d248f8ca6 100644 --- a/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpmanager.c +++ b/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpmanager.c @@ -35,7 +35,7 @@ #include "gstrtpst2022-1-fecdec.h" #include "gstrtpst2022-1-fecenc.h" #include "gstrtphdrext-twcc.h" -#include "gstrtphdrext-rfc6464.h" +#include "gstrtphdrext-clientaudiolevel.h" static gboolean plugin_init (GstPlugin * plugin) @@ -56,7 +56,7 @@ plugin_init (GstPlugin * plugin) ret |= GST_ELEMENT_REGISTER (rtpst2022_1_fecdec, plugin); ret |= GST_ELEMENT_REGISTER (rtpst2022_1_fecenc, plugin); ret |= GST_ELEMENT_REGISTER (rtphdrexttwcc, plugin); - ret |= GST_ELEMENT_REGISTER (rtphdrextrfc6464, plugin); + ret |= GST_ELEMENT_REGISTER (rtphdrextclientaudiolevel, plugin); return ret; } diff --git a/subprojects/gst-plugins-good/gst/rtpmanager/meson.build b/subprojects/gst-plugins-good/gst/rtpmanager/meson.build index 6f8aee2871..186dcc5ef8 100644 --- a/subprojects/gst-plugins-good/gst/rtpmanager/meson.build +++ b/subprojects/gst-plugins-good/gst/rtpmanager/meson.build @@ -4,7 +4,7 @@ rtpmanager_sources = [ 'gstrtpdtmfmux.c', 'gstrtpjitterbuffer.c', 'gstrtphdrext-twcc.c', - 'gstrtphdrext-rfc6464.c', + 'gstrtphdrext-clientaudiolevel.c', 'gstrtpmux.c', 'gstrtpptdemux.c', 'gstrtprtxqueue.c', diff --git a/subprojects/gst-plugins-good/tests/check/elements/rtphdrextrfc6464.c b/subprojects/gst-plugins-good/tests/check/elements/rtphdrextclientaudiolevel.c similarity index 91% rename from subprojects/gst-plugins-good/tests/check/elements/rtphdrextrfc6464.c rename to subprojects/gst-plugins-good/tests/check/elements/rtphdrextclientaudiolevel.c index ae33107bb7..1793ce5ec9 100644 --- a/subprojects/gst-plugins-good/tests/check/elements/rtphdrextrfc6464.c +++ b/subprojects/gst-plugins-good/tests/check/elements/rtphdrextclientaudiolevel.c @@ -98,7 +98,7 @@ check_caps (GstRTPHeaderExtension * ext, gboolean vad) gst_caps_unref (caps); } -GST_START_TEST (rtprfc6464_sdp) +GST_START_TEST (rtphdrext_client_audio_level_sdp) { GstRTPHeaderExtension *ext; GstCaps *caps; @@ -140,7 +140,7 @@ GST_START_TEST (rtprfc6464_sdp) GST_END_TEST; -GST_START_TEST (rtprfc6464_one_byte) +GST_START_TEST (rtphdrext_client_audio_level_one_byte) { GstRTPHeaderExtension *ext; GstRTPHeaderExtensionFlags flags; @@ -188,7 +188,7 @@ GST_START_TEST (rtprfc6464_one_byte) GST_END_TEST; -GST_START_TEST (rtprfc6464_two_bytes) +GST_START_TEST (rtphdrext_client_audio_level_two_bytes) { GstRTPHeaderExtension *ext; GstRTPHeaderExtensionFlags flags; @@ -236,7 +236,7 @@ GST_START_TEST (rtprfc6464_two_bytes) GST_END_TEST; -GST_START_TEST (rtprfc6464_no_meta) +GST_START_TEST (rtphdrext_client_audio_level_no_meta) { GstRTPHeaderExtension *ext; GstBuffer *buffer; @@ -271,7 +271,7 @@ GST_START_TEST (rtprfc6464_no_meta) GST_END_TEST; -GST_START_TEST (rtprfc6464_payloader_depayloader) +GST_START_TEST (rtphdrext_client_audio_level_payloader_depayloader) { GstHarness *h; GstBuffer *b; @@ -304,7 +304,7 @@ GST_START_TEST (rtprfc6464_payloader_depayloader) GST_END_TEST; -GST_START_TEST (rtprfc6464_payloader_api) +GST_START_TEST (rtphdrext_client_audio_level_payloader_api) { GstHarness *h; GstRTPHeaderExtension *ext; @@ -350,21 +350,21 @@ GST_END_TEST; static Suite * -rtprfc6464_suite (void) +rtphdrext_client_audio_level_suite (void) { - Suite *s = suite_create ("rtprfc6464"); + Suite *s = suite_create ("rtphdrext_client_audio_level"); TCase *tc_chain = tcase_create ("general"); suite_add_tcase (s, tc_chain); - tcase_add_test (tc_chain, rtprfc6464_sdp); - tcase_add_test (tc_chain, rtprfc6464_one_byte); - tcase_add_test (tc_chain, rtprfc6464_two_bytes); - tcase_add_test (tc_chain, rtprfc6464_no_meta); - tcase_add_test (tc_chain, rtprfc6464_payloader_depayloader); - tcase_add_test (tc_chain, rtprfc6464_payloader_api); + tcase_add_test (tc_chain, rtphdrext_client_audio_level_sdp); + tcase_add_test (tc_chain, rtphdrext_client_audio_level_one_byte); + tcase_add_test (tc_chain, rtphdrext_client_audio_level_two_bytes); + tcase_add_test (tc_chain, rtphdrext_client_audio_level_no_meta); + tcase_add_test (tc_chain, rtphdrext_client_audio_level_payloader_depayloader); + tcase_add_test (tc_chain, rtphdrext_client_audio_level_payloader_api); return s; } -GST_CHECK_MAIN (rtprfc6464) +GST_CHECK_MAIN (rtphdrext_client_audio_level) diff --git a/subprojects/gst-plugins-good/tests/check/meson.build b/subprojects/gst-plugins-good/tests/check/meson.build index 2f84fe61ca..4460aac722 100644 --- a/subprojects/gst-plugins-good/tests/check/meson.build +++ b/subprojects/gst-plugins-good/tests/check/meson.build @@ -73,7 +73,7 @@ good_tests = [ [ 'elements/rtpbin_buffer_list' ], [ 'elements/rtpcollision' ], [ 'elements/rtpfunnel' ], - [ 'elements/rtphdrextrfc6464', false, [gstsdp_dep, gstaudio_dep] ], + [ 'elements/rtphdrextclientaudiolevel', false, [gstsdp_dep, gstaudio_dep] ], [ 'elements/rtpjitterbuffer' ], [ 'elements/rtpjpeg' ],