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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>
370 lines
10 KiB
C
370 lines
10 KiB
C
/* GStreamer
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*
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* unit test for RTP RFC 6464 Header Extensions
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*
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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 details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <gst/check/gstcheck.h>
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#include <gst/rtp/rtp.h>
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#include <gst/sdp/gstsdpmessage.h>
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#include <gst/audio/audio.h>
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#include <gst/check/gstharness.h>
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#define URN "urn:ietf:params:rtp-hdrext:ssrc-audio-level"
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#define SDP "v=0\r\n" \
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"o=- 123456 2 IN IP4 127.0.0.1 \r\n" \
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"s=-\r\n" \
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"t=0 0\r\n" \
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"a=maxptime:60\r\n" \
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"a=sendrecv\r\n" \
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"m=audio 55815 RTP/SAVPF 100\r\n" \
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"c=IN IP4 1.1.1.1\r\n" \
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"a=rtpmap:100 opus/48000/2\r\n"
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#define SDP_NO_VAD SDP \
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"a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\n"
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#define SDP_VAD_ON SDP \
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"a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=on\r\n"
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#define SDP_VAD_OFF SDP \
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"a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=off\r\n"
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#define SDP_VAD_WRONG SDP \
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"a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=badger\r\n"
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static GstCaps *
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create_caps (const gchar * sdp)
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{
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GstSDPMessage *message;
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glong length = -1;
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const GstSDPMedia *media;
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GstCaps *caps;
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gst_sdp_message_new (&message);
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gst_sdp_message_parse_buffer ((guint8 *) sdp, length, message);
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media = gst_sdp_message_get_media (message, 0);
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fail_unless (media != NULL);
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caps = gst_sdp_media_get_caps_from_media (media, 100);
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gst_sdp_media_attributes_to_caps (media, caps);
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gst_sdp_message_free (message);
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return caps;
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}
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static void
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check_caps (GstRTPHeaderExtension * ext, gboolean vad)
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{
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GstCaps *caps;
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GstStructure *s;
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const GValue *arr, *val;
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caps = gst_caps_new_empty_simple ("application/x-rtp");
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fail_unless (gst_rtp_header_extension_set_caps_from_attributes (ext, caps));
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s = gst_caps_get_structure (caps, 0);
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arr = gst_structure_get_value (s, "extmap-1");
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fail_unless (arr != NULL);
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fail_unless (GST_VALUE_HOLDS_ARRAY (arr));
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fail_unless (gst_value_array_get_size (arr) == 3);
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val = gst_value_array_get_value (arr, 0);
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fail_unless_equals_string (g_value_get_string (val), "");
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val = gst_value_array_get_value (arr, 1);
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fail_unless_equals_string (g_value_get_string (val), URN);
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val = gst_value_array_get_value (arr, 2);
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if (vad) {
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fail_unless_equals_string (g_value_get_string (val), "vad=on");
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} else {
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fail_unless_equals_string (g_value_get_string (val), "vad=off");
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}
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gst_caps_unref (caps);
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}
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GST_START_TEST (rtphdrext_client_audio_level_sdp)
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{
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GstRTPHeaderExtension *ext;
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GstCaps *caps;
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gboolean vad = FALSE;
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ext = gst_rtp_header_extension_create_from_uri (URN);
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fail_unless (ext != NULL);
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gst_rtp_header_extension_set_id (ext, 1);
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/* vad default to on */
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caps = create_caps (SDP_NO_VAD);
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fail_unless (gst_rtp_header_extension_set_attributes_from_caps (ext, caps));
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gst_caps_unref (caps);
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g_object_get (ext, "vad", &vad, NULL);
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fail_unless (vad);
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check_caps (ext, TRUE);
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/* vad is disabled */
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caps = create_caps (SDP_VAD_OFF);
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fail_unless (gst_rtp_header_extension_set_attributes_from_caps (ext, caps));
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gst_caps_unref (caps);
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g_object_get (ext, "vad", &vad, NULL);
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fail_if (vad);
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/* vad is enabled */
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caps = create_caps (SDP_VAD_ON);
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fail_unless (gst_rtp_header_extension_set_attributes_from_caps (ext, caps));
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gst_caps_unref (caps);
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g_object_get (ext, "vad", &vad, NULL);
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fail_unless (vad);
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/* invalid vad */
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caps = create_caps (SDP_VAD_WRONG);
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fail_if (gst_rtp_header_extension_set_attributes_from_caps (ext, caps));
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gst_caps_unref (caps);
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gst_object_unref (ext);
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}
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GST_END_TEST;
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GST_START_TEST (rtphdrext_client_audio_level_one_byte)
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{
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GstRTPHeaderExtension *ext;
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GstRTPHeaderExtensionFlags flags;
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GstBuffer *buffer;
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guint8 *data;
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gsize size, written;
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GstAudioLevelMeta *meta;
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guint8 level = 12;
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gboolean voice = TRUE;
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ext = gst_rtp_header_extension_create_from_uri (URN);
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fail_unless (ext != NULL);
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gst_rtp_header_extension_set_id (ext, 1);
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flags = gst_rtp_header_extension_get_supported_flags (ext);
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fail_unless (flags & GST_RTP_HEADER_EXTENSION_ONE_BYTE);
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buffer = gst_buffer_new ();
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meta = gst_buffer_add_audio_level_meta (buffer, level, voice);
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size = gst_rtp_header_extension_get_max_size (ext, buffer);
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fail_unless (size > 0);
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data = g_malloc0 (size);
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fail_unless (data != NULL);
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/* Write extension */
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written =
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gst_rtp_header_extension_write (ext, buffer,
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GST_RTP_HEADER_EXTENSION_ONE_BYTE, buffer, data, size);
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fail_unless (written == 1);
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/* Read it back */
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fail_unless (gst_buffer_remove_meta (buffer, (GstMeta *) meta));
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fail_unless (gst_rtp_header_extension_read (ext,
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GST_RTP_HEADER_EXTENSION_ONE_BYTE, data, size, buffer));
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meta = gst_buffer_get_audio_level_meta (buffer);
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fail_unless (meta != NULL);
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fail_unless_equals_int (meta->level, level);
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fail_unless (meta->voice_activity == voice);
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g_free (data);
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gst_buffer_unref (buffer);
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gst_object_unref (ext);
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}
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GST_END_TEST;
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GST_START_TEST (rtphdrext_client_audio_level_two_bytes)
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{
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GstRTPHeaderExtension *ext;
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GstRTPHeaderExtensionFlags flags;
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GstBuffer *buffer;
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guint8 *data;
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gsize size, written;
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GstAudioLevelMeta *meta;
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guint8 level = 12;
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gboolean voice = TRUE;
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ext = gst_rtp_header_extension_create_from_uri (URN);
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fail_unless (ext != NULL);
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gst_rtp_header_extension_set_id (ext, 1);
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flags = gst_rtp_header_extension_get_supported_flags (ext);
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fail_unless (flags & GST_RTP_HEADER_EXTENSION_TWO_BYTE);
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buffer = gst_buffer_new ();
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meta = gst_buffer_add_audio_level_meta (buffer, level, voice);
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size = gst_rtp_header_extension_get_max_size (ext, buffer);
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fail_unless (size > 0);
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data = g_malloc0 (size);
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fail_unless (data != NULL);
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/* Write extension */
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written =
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gst_rtp_header_extension_write (ext, buffer,
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GST_RTP_HEADER_EXTENSION_TWO_BYTE, buffer, data, size);
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fail_unless (written == 2);
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/* Read it back */
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fail_unless (gst_buffer_remove_meta (buffer, (GstMeta *) meta));
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fail_unless (gst_rtp_header_extension_read (ext,
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GST_RTP_HEADER_EXTENSION_TWO_BYTE, data, size, buffer));
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meta = gst_buffer_get_audio_level_meta (buffer);
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fail_unless (meta != NULL);
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fail_unless_equals_int (meta->level, level);
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fail_unless (meta->voice_activity == voice);
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g_free (data);
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gst_buffer_unref (buffer);
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gst_object_unref (ext);
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}
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GST_END_TEST;
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GST_START_TEST (rtphdrext_client_audio_level_no_meta)
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{
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GstRTPHeaderExtension *ext;
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GstBuffer *buffer;
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guint8 *data;
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gsize size, written;
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ext = gst_rtp_header_extension_create_from_uri (URN);
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fail_unless (ext != NULL);
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gst_rtp_header_extension_set_id (ext, 1);
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buffer = gst_buffer_new ();
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size = gst_rtp_header_extension_get_max_size (ext, buffer);
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fail_unless (size > 0);
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data = g_malloc0 (size);
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fail_unless (data != NULL);
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written =
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gst_rtp_header_extension_write (ext, buffer,
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GST_RTP_HEADER_EXTENSION_ONE_BYTE, buffer, data, size);
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fail_unless (written == 0);
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written =
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gst_rtp_header_extension_write (ext, buffer,
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GST_RTP_HEADER_EXTENSION_TWO_BYTE, buffer, data, size);
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fail_unless (written == 0);
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g_free (data);
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gst_buffer_unref (buffer);
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gst_object_unref (ext);
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}
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GST_END_TEST;
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GST_START_TEST (rtphdrext_client_audio_level_payloader_depayloader)
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{
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GstHarness *h;
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GstBuffer *b;
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GstFlowReturn fret;
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GstAudioLevelMeta *meta;
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h = gst_harness_new_parse ("rtpL16pay ! "
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"application/x-rtp, extmap-1=(string)< \"\", " URN " , \"vad=on\" >"
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" ! rtpL16depay");
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gst_harness_set_src_caps_str (h, "audio/x-raw, rate=44100, channels=1,"
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" layout=interleaved, format=S16BE");
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b = gst_buffer_new_allocate (NULL, 100, NULL);
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gst_buffer_add_audio_level_meta (b, 12, TRUE);
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fret = gst_harness_push (h, b);
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fail_unless (fret == GST_FLOW_OK);
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b = gst_harness_pull (h);
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meta = gst_buffer_get_audio_level_meta (b);
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fail_unless (meta != NULL);
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fail_unless (meta->level == 12);
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fail_unless (meta->voice_activity == TRUE);
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gst_buffer_unref (b);
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gst_harness_teardown (h);
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}
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GST_END_TEST;
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GST_START_TEST (rtphdrext_client_audio_level_payloader_api)
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{
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GstHarness *h;
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GstRTPHeaderExtension *ext;
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GstBuffer *b;
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GstFlowReturn fret;
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GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
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guint8 *data;
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guint size;
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guint8 level;
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gboolean voice_activity;
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h = gst_harness_new ("rtpL16pay");
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gst_harness_set_src_caps_str (h, "audio/x-raw, rate=44100, channels=1,"
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" layout=interleaved, format=S16BE");
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ext = gst_rtp_header_extension_create_from_uri (URN);
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gst_rtp_header_extension_set_id (ext, 2);
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fail_unless (ext);
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g_signal_emit_by_name (h->element, "add-extension", ext);
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b = gst_buffer_new_allocate (NULL, 100, NULL);
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gst_buffer_add_audio_level_meta (b, 12, TRUE);
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fret = gst_harness_push (h, b);
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fail_unless (fret == GST_FLOW_OK);
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b = gst_harness_pull (h);
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fail_unless (gst_rtp_buffer_map (b, GST_MAP_READ, &rtp));
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fail_unless (gst_rtp_buffer_get_extension_onebyte_header (&rtp, 2, 0,
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(gpointer *) & data, &size));
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fail_unless (size == 1);
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level = data[0] & 0x7F;
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voice_activity = (data[0] & 0x80) >> 7;
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fail_unless (level == 12);
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fail_unless (voice_activity == TRUE);
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gst_rtp_buffer_unmap (&rtp);
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gst_buffer_unref (b);
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gst_object_unref (ext);
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gst_harness_teardown (h);
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}
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GST_END_TEST;
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static Suite *
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rtphdrext_client_audio_level_suite (void)
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{
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Suite *s = suite_create ("rtphdrext_client_audio_level");
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TCase *tc_chain = tcase_create ("general");
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suite_add_tcase (s, tc_chain);
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tcase_add_test (tc_chain, rtphdrext_client_audio_level_sdp);
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tcase_add_test (tc_chain, rtphdrext_client_audio_level_one_byte);
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tcase_add_test (tc_chain, rtphdrext_client_audio_level_two_bytes);
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tcase_add_test (tc_chain, rtphdrext_client_audio_level_no_meta);
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tcase_add_test (tc_chain, rtphdrext_client_audio_level_payloader_depayloader);
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tcase_add_test (tc_chain, rtphdrext_client_audio_level_payloader_api);
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return s;
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}
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GST_CHECK_MAIN (rtphdrext_client_audio_level)
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