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rtp-payloading: Add new test for Vorbis renegotiation

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Check if encoding, payloading, depayloading and decoding works if the
stream configuration (and thus the headers) change.
This commit is contained in:
Sebastian Dröge 2017-02-27 19:25:35 +02:00
parent f44314c029
commit eefcdc9ee1
2 changed files with 94 additions and 0 deletions
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3
tests/check/Makefile.am
View file

@ -483,6 +483,9 @@ elements_rglimiter_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION
elements_rgvolume_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS) elements_rgvolume_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
elements_rgvolume_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) $(LDADD) elements_rgvolume_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) $(LDADD)
elements_rtp_payloading_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
elements_rtp_payloading_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) $(LDADD)
elements_spectrum_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS) elements_spectrum_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
elements_spectrum_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) $(LDADD) elements_spectrum_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) $(LDADD)

91
tests/check/elements/rtp-payloading.c
View file

@ -19,6 +19,7 @@
*/ */
#include <gst/check/gstcheck.h> #include <gst/check/gstcheck.h>
#include <gst/check/gstharness.h> #include <gst/check/gstharness.h>
#include <gst/audio/audio.h>
#include <stdlib.h> #include <stdlib.h>
#include <unistd.h> #include <unistd.h>
@ -1329,6 +1330,95 @@ GST_START_TEST (rtp_gst_custom_event)
GST_END_TEST; GST_END_TEST;
GST_START_TEST (rtp_vorbis_renegotiate)
{
GstElement *pipeline;
GstElement *enc, *pay, *depay, *dec, *sink;
GstPad *sinkpad, *srcpad;
GstCaps *templcaps, *caps, *filter, *srccaps;
GstSegment segment;
GstBuffer *buffer;
GstMapInfo map;
GstAudioInfo info;
pipeline = gst_pipeline_new (NULL);
enc = gst_element_factory_make ("vorbisenc", NULL);
pay = gst_element_factory_make ("rtpvorbispay", NULL);
depay = gst_element_factory_make ("rtpvorbisdepay", NULL);
dec = gst_element_factory_make ("vorbisdec", NULL);
sink = gst_element_factory_make ("fakesink", NULL);
g_object_set (sink, "async", FALSE, NULL);
gst_bin_add_many (GST_BIN (pipeline), enc, pay, depay, dec, sink, NULL);
fail_unless (gst_element_link_many (enc, pay, depay, dec, sink, NULL));
fail_unless_equals_int (gst_element_set_state (pipeline, GST_STATE_PLAYING),
GST_STATE_CHANGE_SUCCESS);
sinkpad = gst_element_get_static_pad (enc, "sink");
srcpad = gst_element_get_static_pad (dec, "src");
templcaps = gst_pad_get_pad_template_caps (sinkpad);
filter =
gst_caps_new_simple ("audio/x-raw", "channels", G_TYPE_INT, 2, "rate",
G_TYPE_INT, 44100, NULL);
caps = gst_caps_intersect (templcaps, filter);
caps = gst_caps_fixate (caps);
gst_segment_init (&segment, GST_FORMAT_TIME);
fail_unless (gst_pad_send_event (sinkpad,
gst_event_new_stream_start ("test")));
fail_unless (gst_pad_send_event (sinkpad, gst_event_new_caps (caps)));
fail_unless (gst_pad_send_event (sinkpad, gst_event_new_segment (&segment)));
gst_audio_info_from_caps (&info, caps);
buffer = gst_buffer_new_and_alloc (44100 * info.bpf);
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
gst_audio_format_fill_silence (info.finfo, map.data, map.size);
gst_buffer_unmap (buffer, &map);
GST_BUFFER_PTS (buffer) = 0;
GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
fail_unless_equals_int (gst_pad_chain (sinkpad, buffer), GST_FLOW_OK);
srccaps = gst_pad_get_current_caps (srcpad);
fail_unless (gst_caps_can_intersect (srccaps, caps));
gst_caps_unref (srccaps);
gst_caps_unref (caps);
gst_caps_unref (filter);
filter =
gst_caps_new_simple ("audio/x-raw", "channels", G_TYPE_INT, 2, "rate",
G_TYPE_INT, 48000, NULL);
caps = gst_caps_intersect (templcaps, filter);
caps = gst_caps_fixate (caps);
fail_unless (gst_pad_send_event (sinkpad, gst_event_new_caps (caps)));
gst_audio_info_from_caps (&info, caps);
buffer = gst_buffer_new_and_alloc (48000 * info.bpf);
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
gst_audio_format_fill_silence (info.finfo, map.data, map.size);
gst_buffer_unmap (buffer, &map);
GST_BUFFER_PTS (buffer) = 0;
GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
fail_unless_equals_int (gst_pad_chain (sinkpad, buffer), GST_FLOW_OK);
srccaps = gst_pad_get_current_caps (srcpad);
fail_unless (gst_caps_can_intersect (srccaps, caps));
gst_caps_unref (srccaps);
gst_caps_unref (caps);
gst_caps_unref (filter);
gst_caps_unref (templcaps);
gst_object_unref (sinkpad);
gst_object_unref (srcpad);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
}
GST_END_TEST;
/* /*
* Creates the test suite. * Creates the test suite.
* *
@ -1388,6 +1478,7 @@ rtp_payloading_suite (void)
tcase_add_loop_test (tc_chain, rtp_jpeg_packet_loss, 0, 7); tcase_add_loop_test (tc_chain, rtp_jpeg_packet_loss, 0, 7);
tcase_add_test (tc_chain, rtp_g729); tcase_add_test (tc_chain, rtp_g729);
tcase_add_test (tc_chain, rtp_gst_custom_event); tcase_add_test (tc_chain, rtp_gst_custom_event);
tcase_add_test (tc_chain, rtp_vorbis_renegotiate);
return s; return s;
} }