gstreamer/subprojects/gst-plugins-good/tests/check/elements/rtphdrextclientaudiolevel.c

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