gstreamer/tests/check/elements/flvmux.c
Yeongjin Jeong 830db205f6 tests: flvmux: Instead of using the testclock, just send eos event for drain
When using the testclock for determining clock in test, it is sometimes observed
that the clock entry is not registered in time by the aggregator. So deadlock occurs
between the aggregator and the test thread.
2020-03-02 01:37:27 +09:00

1176 lines
40 KiB
C

/* GStreamer unit tests for flvmux
*
* Copyright (C) 2009 Tim-Philipp Müller <tim centricular net>
* Copyright (C) 2016 Havard Graff <havard@pexip.com>
* Copyright (C) 2016 David Buchmann <david@pexip.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.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#ifdef HAVE_VALGRIND
# include <valgrind/valgrind.h>
#endif
#include <gst/check/gstcheck.h>
#include <gst/check/gstharness.h>
#include <gst/gst.h>
static GstBusSyncReply
error_cb (GstBus * bus, GstMessage * msg, gpointer user_data)
{
if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR) {
GError *err = NULL;
gchar *dbg = NULL;
gst_message_parse_error (msg, &err, &dbg);
g_error ("ERROR: %s\n%s\n", err->message, dbg);
}
return GST_BUS_PASS;
}
static void
handoff_cb (GstElement * element, GstBuffer * buf, GstPad * pad,
gint * p_counter)
{
*p_counter += 1;
GST_LOG ("counter = %d", *p_counter);
}
static void
mux_pcm_audio (guint num_buffers, guint repeat)
{
GstElement *src, *sink, *flvmux, *conv, *pipeline;
GstPad *sinkpad, *srcpad;
gint counter;
GST_LOG ("num_buffers = %u", num_buffers);
pipeline = gst_pipeline_new ("pipeline");
fail_unless (pipeline != NULL, "Failed to create pipeline!");
/* kids, don't use a sync handler for this at home, really; we do because
* we just want to abort and nothing else */
gst_bus_set_sync_handler (GST_ELEMENT_BUS (pipeline), error_cb, NULL, NULL);
src = gst_element_factory_make ("audiotestsrc", "audiotestsrc");
fail_unless (src != NULL, "Failed to create 'audiotestsrc' element!");
g_object_set (src, "num-buffers", num_buffers, NULL);
conv = gst_element_factory_make ("audioconvert", "audioconvert");
fail_unless (conv != NULL, "Failed to create 'audioconvert' element!");
flvmux = gst_element_factory_make ("flvmux", "flvmux");
fail_unless (flvmux != NULL, "Failed to create 'flvmux' element!");
sink = gst_element_factory_make ("fakesink", "fakesink");
fail_unless (sink != NULL, "Failed to create 'fakesink' element!");
g_object_set (sink, "signal-handoffs", TRUE, NULL);
g_signal_connect (sink, "handoff", G_CALLBACK (handoff_cb), &counter);
gst_bin_add_many (GST_BIN (pipeline), src, conv, flvmux, sink, NULL);
fail_unless (gst_element_link (src, conv));
fail_unless (gst_element_link (flvmux, sink));
/* now link the elements */
sinkpad = gst_element_get_request_pad (flvmux, "audio");
fail_unless (sinkpad != NULL, "Could not get audio request pad");
srcpad = gst_element_get_static_pad (conv, "src");
fail_unless (srcpad != NULL, "Could not get audioconvert's source pad");
fail_unless_equals_int (gst_pad_link (srcpad, sinkpad), GST_PAD_LINK_OK);
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
do {
GstStateChangeReturn state_ret;
GstMessage *msg;
GST_LOG ("repeat=%d", repeat);
counter = 0;
state_ret = gst_element_set_state (pipeline, GST_STATE_PAUSED);
fail_unless (state_ret != GST_STATE_CHANGE_FAILURE);
if (state_ret == GST_STATE_CHANGE_ASYNC) {
GST_LOG ("waiting for pipeline to reach PAUSED state");
state_ret = gst_element_get_state (pipeline, NULL, NULL, -1);
fail_unless_equals_int (state_ret, GST_STATE_CHANGE_SUCCESS);
}
GST_LOG ("PAUSED, let's do the rest of it");
state_ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
fail_unless (state_ret != GST_STATE_CHANGE_FAILURE);
msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
fail_unless (msg != NULL, "Expected EOS message on bus!");
GST_LOG ("EOS");
gst_message_unref (msg);
/* should have some output */
fail_unless (counter > 2);
fail_unless_equals_int (gst_element_set_state (pipeline, GST_STATE_NULL),
GST_STATE_CHANGE_SUCCESS);
/* repeat = test re-usability */
--repeat;
} while (repeat > 0);
gst_object_unref (pipeline);
}
GST_START_TEST (test_index_writing)
{
/* note: there's a magic 128 value in flvmux when doing index writing */
mux_pcm_audio (__i__ * 33 + 1, 2);
}
GST_END_TEST;
static GstBuffer *
create_buffer (guint8 * data, gsize size,
GstClockTime timestamp, GstClockTime duration)
{
GstBuffer *buf = gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY,
data, size, 0, size, NULL, NULL);
GST_BUFFER_PTS (buf) = timestamp;
GST_BUFFER_DTS (buf) = timestamp;
GST_BUFFER_DURATION (buf) = duration;
GST_BUFFER_OFFSET (buf) = 0;
GST_BUFFER_OFFSET_END (buf) = 0;
return buf;
}
guint8 speex_hdr0[] = {
0x53, 0x70, 0x65, 0x65, 0x78, 0x20, 0x20, 0x20,
0x31, 0x2e, 0x32, 0x72, 0x63, 0x31, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00,
0x50, 0x00, 0x00, 0x00, 0x80, 0x3e, 0x00, 0x00,
0x01, 0x00, 0x00, 0x00, 0x04, 0x00, 0x00, 0x00,
0x01, 0x00, 0x00, 0x00, 0xff, 0xff, 0xff, 0xff,
0x40, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
guint8 speex_hdr1[] = {
0x1f, 0x00, 0x00, 0x00, 0x45, 0x6e, 0x63, 0x6f,
0x64, 0x65, 0x64, 0x20, 0x77, 0x69, 0x74, 0x68,
0x20, 0x47, 0x53, 0x74, 0x72, 0x65, 0x61, 0x6d,
0x65, 0x72, 0x20, 0x53, 0x70, 0x65, 0x65, 0x78,
0x65, 0x6e, 0x63, 0x00, 0x00, 0x00, 0x00, 0x01
};
guint8 speex_buf[] = {
0x36, 0x9d, 0x1b, 0x9a, 0x20, 0x00, 0x01, 0x68,
0xe8, 0xe8, 0xe8, 0xe8, 0xe8, 0xe8, 0xe8, 0x84,
0x00, 0xb4, 0x74, 0x74, 0x74, 0x74, 0x74, 0x74,
0x74, 0x42, 0x00, 0x5a, 0x3a, 0x3a, 0x3a, 0x3a,
0x3a, 0x3a, 0x3a, 0x21, 0x00, 0x2d, 0x1d, 0x1d,
0x1d, 0x1d, 0x1d, 0x1d, 0x1d, 0x1b, 0x3b, 0x60,
0xab, 0xab, 0xab, 0xab, 0xab, 0x0a, 0xba, 0xba,
0xba, 0xba, 0xb0, 0xab, 0xab, 0xab, 0xab, 0xab,
0x0a, 0xba, 0xba, 0xba, 0xba, 0xb7
};
guint8 h264_buf[] = {
0x00, 0x00, 0x00, 0x0b, 0x67, 0x42, 0xc0, 0x0c,
0x95, 0xa7, 0x20, 0x1e, 0x11, 0x08, 0xd4, 0x00,
0x00, 0x00, 0x04, 0x68, 0xce, 0x3c, 0x80, 0x00,
0x00, 0x00, 0x55, 0x65, 0xb8, 0x04, 0x0e, 0x7e,
0x1f, 0x22, 0x60, 0x34, 0x01, 0xe2, 0x00, 0x3c,
0xe1, 0xfc, 0x91, 0x40, 0xa6, 0x9e, 0x07, 0x42,
0x56, 0x44, 0x73, 0x75, 0x40, 0x9f, 0x0c, 0x87,
0x83, 0xc9, 0x52, 0x60, 0x6d, 0xd8, 0x98, 0x01,
0x16, 0xbd, 0x0f, 0xa6, 0xaf, 0x75, 0x83, 0xdd,
0xfa, 0xe7, 0x8f, 0xe3, 0x58, 0x10, 0x0f, 0x5c,
0x18, 0x2f, 0x41, 0x40, 0x23, 0x0b, 0x03, 0x70,
0x00, 0xff, 0xe4, 0xa6, 0x7d, 0x7f, 0x3f, 0x76,
0x01, 0xd0, 0x98, 0x2a, 0x0c, 0xb8, 0x02, 0x32,
0xbc, 0x56, 0xfd, 0x34, 0x4f, 0xcf, 0xfe, 0xa0,
};
GST_START_TEST (test_speex_streamable)
{
GstBuffer *buf;
GstMapInfo map = GST_MAP_INFO_INIT;
GstCaps *caps = gst_caps_new_simple ("audio/x-speex",
"rate", G_TYPE_INT, 16000,
"channels", G_TYPE_INT, 1,
NULL);
const GstClockTime base_time = 123456789;
const GstClockTime duration_ms = 20;
const GstClockTime duration = duration_ms * GST_MSECOND;
GstHarness *h = gst_harness_new_with_padnames ("flvmux", "audio", "src");
gst_harness_set_src_caps (h, caps);
g_object_set (h->element, "streamable", 1, NULL);
/* push speex header0 */
gst_harness_push (h, create_buffer (speex_hdr0,
sizeof (speex_hdr0), base_time, 0));
/* push speex header1 */
gst_harness_push (h, create_buffer (speex_hdr1,
sizeof (speex_hdr1), base_time, 0));
/* push speex data */
gst_harness_push (h, create_buffer (speex_buf,
sizeof (speex_buf), base_time, duration));
/* push speex data 2 */
gst_harness_push (h, create_buffer (speex_buf,
sizeof (speex_buf), base_time + duration, duration));
/* pull out stream-start event */
gst_event_unref (gst_harness_pull_event (h));
/* pull out caps event */
gst_event_unref (gst_harness_pull_event (h));
/* pull out segment event and verify we are using GST_FORMAT_TIME */
{
GstEvent *event = gst_harness_pull_event (h);
const GstSegment *segment;
gst_event_parse_segment (event, &segment);
fail_unless_equals_int (GST_FORMAT_TIME, segment->format);
gst_event_unref (event);
}
/* pull FLV header buffer */
buf = gst_harness_pull (h);
gst_buffer_unref (buf);
/* pull Metadata buffer */
buf = gst_harness_pull (h);
gst_buffer_unref (buf);
/* pull header0 */
buf = gst_harness_pull (h);
fail_unless_equals_uint64 (base_time, GST_BUFFER_PTS (buf));
fail_unless_equals_uint64 (GST_CLOCK_TIME_NONE, GST_BUFFER_DTS (buf));
fail_unless_equals_uint64 (0, GST_BUFFER_DURATION (buf));
gst_buffer_map (buf, &map, GST_MAP_READ);
/* 0x08 means it is audio */
fail_unless_equals_int (0x08, map.data[0]);
/* timestamp should be starting from 0 */
fail_unless_equals_int (0x00, map.data[6]);
/* 0xb2 means Speex, 16000Hz, Mono */
fail_unless_equals_int (0xb2, map.data[11]);
/* verify content is intact */
fail_unless_equals_int (0, memcmp (&map.data[12], speex_hdr0,
sizeof (speex_hdr0)));
gst_buffer_unmap (buf, &map);
gst_buffer_unref (buf);
/* pull header1 */
buf = gst_harness_pull (h);
fail_unless_equals_uint64 (base_time, GST_BUFFER_PTS (buf));
fail_unless_equals_uint64 (GST_CLOCK_TIME_NONE, GST_BUFFER_DTS (buf));
fail_unless_equals_uint64 (0, GST_BUFFER_DURATION (buf));
gst_buffer_map (buf, &map, GST_MAP_READ);
/* 0x08 means it is audio */
fail_unless_equals_int (0x08, map.data[0]);
/* timestamp should be starting from 0 */
fail_unless_equals_int (0x00, map.data[6]);
/* 0xb2 means Speex, 16000Hz, Mono */
fail_unless_equals_int (0xb2, map.data[11]);
/* verify content is intact */
fail_unless_equals_int (0, memcmp (&map.data[12], speex_hdr1,
sizeof (speex_hdr1)));
gst_buffer_unmap (buf, &map);
gst_buffer_unref (buf);
/* pull data */
buf = gst_harness_pull (h);
fail_unless_equals_uint64 (base_time, GST_BUFFER_PTS (buf));
fail_unless_equals_uint64 (GST_CLOCK_TIME_NONE, GST_BUFFER_DTS (buf));
fail_unless_equals_uint64 (duration, GST_BUFFER_DURATION (buf));
fail_unless_equals_uint64 (GST_BUFFER_OFFSET_NONE, GST_BUFFER_OFFSET (buf));
fail_unless_equals_uint64 (GST_BUFFER_OFFSET_NONE,
GST_BUFFER_OFFSET_END (buf));
gst_buffer_map (buf, &map, GST_MAP_READ);
/* 0x08 means it is audio */
fail_unless_equals_int (0x08, map.data[0]);
/* timestamp should be starting from 0 */
fail_unless_equals_int (0x00, map.data[6]);
/* 0xb2 means Speex, 16000Hz, Mono */
fail_unless_equals_int (0xb2, map.data[11]);
/* verify content is intact */
fail_unless_equals_int (0, memcmp (&map.data[12], speex_buf,
sizeof (speex_buf)));
gst_buffer_unmap (buf, &map);
gst_buffer_unref (buf);
/* pull data */
buf = gst_harness_pull (h);
fail_unless_equals_uint64 (base_time + duration, GST_BUFFER_PTS (buf));
fail_unless_equals_uint64 (GST_CLOCK_TIME_NONE, GST_BUFFER_DTS (buf));
fail_unless_equals_uint64 (duration, GST_BUFFER_DURATION (buf));
fail_unless_equals_uint64 (GST_BUFFER_OFFSET_NONE, GST_BUFFER_OFFSET (buf));
fail_unless_equals_uint64 (GST_BUFFER_OFFSET_NONE,
GST_BUFFER_OFFSET_END (buf));
gst_buffer_map (buf, &map, GST_MAP_READ);
/* 0x08 means it is audio */
fail_unless_equals_int (0x08, map.data[0]);
/* timestamp should reflect the duration_ms */
fail_unless_equals_int (duration_ms, map.data[6]);
/* 0xb2 means Speex, 16000Hz, Mono */
fail_unless_equals_int (0xb2, map.data[11]);
/* verify content is intact */
fail_unless_equals_int (0, memcmp (&map.data[12], speex_buf,
sizeof (speex_buf)));
gst_buffer_unmap (buf, &map);
gst_buffer_unref (buf);
gst_harness_teardown (h);
}
GST_END_TEST;
static void
check_buf_type_timestamp (GstBuffer * buf, gint packet_type, gint timestamp)
{
GstMapInfo map = GST_MAP_INFO_INIT;
gst_buffer_map (buf, &map, GST_MAP_READ);
fail_unless_equals_int (packet_type, map.data[0]);
fail_unless_equals_int (timestamp, map.data[6]);
gst_buffer_unmap (buf, &map);
gst_buffer_unref (buf);
}
static const gint AUDIO = 0x08;
static const gint VIDEO = 0x09;
GST_START_TEST (test_increasing_timestamp_when_pts_none)
{
gint timestamp = 3;
GstClockTime base_time = 42 * GST_SECOND;
GstPad *audio_sink, *video_sink, *audio_src, *video_src;
GstHarness *h, *audio, *video, *audio_q, *video_q;
GstCaps *audio_caps, *video_caps;
GstBuffer *buf;
h = gst_harness_new_with_padnames ("flvmux", NULL, "src");
audio = gst_harness_new_with_element (h->element, "audio", NULL);
video = gst_harness_new_with_element (h->element, "video", NULL);
audio_q = gst_harness_new ("queue");
video_q = gst_harness_new ("queue");
audio_sink = GST_PAD_PEER (audio->srcpad);
video_sink = GST_PAD_PEER (video->srcpad);
audio_src = GST_PAD_PEER (audio_q->sinkpad);
video_src = GST_PAD_PEER (video_q->sinkpad);
gst_pad_unlink (audio->srcpad, audio_sink);
gst_pad_unlink (video->srcpad, video_sink);
gst_pad_unlink (audio_src, audio_q->sinkpad);
gst_pad_unlink (video_src, video_q->sinkpad);
gst_pad_link (audio_src, audio_sink);
gst_pad_link (video_src, video_sink);
audio_caps = gst_caps_new_simple ("audio/x-speex",
"rate", G_TYPE_INT, 16000, "channels", G_TYPE_INT, 1, NULL);
gst_harness_set_src_caps (audio_q, audio_caps);
video_caps = gst_caps_new_simple ("video/x-h264",
"stream-format", G_TYPE_STRING, "avc", NULL);
gst_harness_set_src_caps (video_q, video_caps);
/* Push audio + video + audio with increasing DTS, but PTS for video is
* GST_CLOCK_TIME_NONE
*/
buf = gst_buffer_new ();
GST_BUFFER_DTS (buf) = timestamp * GST_MSECOND + base_time;
GST_BUFFER_PTS (buf) = timestamp * GST_MSECOND + base_time;
gst_harness_push (audio_q, buf);
buf = gst_buffer_new ();
GST_BUFFER_DTS (buf) = (timestamp + 1) * GST_MSECOND + base_time;
GST_BUFFER_PTS (buf) = GST_CLOCK_TIME_NONE;
gst_harness_push (video_q, buf);
buf = gst_buffer_new ();
GST_BUFFER_DTS (buf) = (timestamp + 2) * GST_MSECOND + base_time;
GST_BUFFER_PTS (buf) = (timestamp + 2) * GST_MSECOND + base_time;
gst_harness_push (audio_q, buf);
/* Pull two metadata packets out */
gst_buffer_unref (gst_harness_pull (h));
gst_buffer_unref (gst_harness_pull (h));
/* Check that we receive the packets in monotonically increasing order and
* that their timestamps are correct (should start at 0)
*/
buf = gst_harness_pull (h);
check_buf_type_timestamp (buf, AUDIO, 0);
buf = gst_harness_pull (h);
check_buf_type_timestamp (buf, VIDEO, 1);
/* teardown */
gst_harness_teardown (h);
gst_harness_teardown (audio);
gst_harness_teardown (video);
gst_harness_teardown (audio_q);
gst_harness_teardown (video_q);
}
GST_END_TEST;
typedef struct
{
GstHarness *a_sink;
GstHarness *v_sink;
} DemuxHarnesses;
static void
flvdemux_pad_added (GstElement * flvdemux, GstPad * srcpad, DemuxHarnesses * h)
{
GstCaps *caps = gst_pad_get_current_caps (srcpad);
const gchar *name = gst_structure_get_name (gst_caps_get_structure (caps, 0));
if (h->a_sink && g_ascii_strncasecmp ("audio", name, 5) == 0)
gst_harness_add_element_src_pad (h->a_sink, srcpad);
else if (h->v_sink && g_ascii_strncasecmp ("video", name, 5) == 0)
gst_harness_add_element_src_pad (h->v_sink, srcpad);
else
ck_abort_msg ("Unexpected demux pad: %s", GST_STR_NULL (name));
gst_caps_unref (caps);
}
GST_START_TEST (test_video_caps_late)
{
GstHarness *mux = gst_harness_new_with_padnames ("flvmux", NULL, "src");
GstHarness *a_src =
gst_harness_new_with_element (mux->element, "audio", NULL);
GstHarness *v_src =
gst_harness_new_with_element (mux->element, "video", NULL);
GstHarness *demux = gst_harness_new_with_padnames ("flvdemux", "sink", NULL);
GstHarness *a_sink =
gst_harness_new_with_element (demux->element, NULL, NULL);
GstHarness *v_sink =
gst_harness_new_with_element (demux->element, NULL, NULL);
DemuxHarnesses harnesses = { a_sink, v_sink };
GstTestClock *tclock;
g_object_set (mux->element, "streamable", TRUE,
"latency", G_GUINT64_CONSTANT (1), NULL);
gst_harness_use_testclock (mux);
g_signal_connect (demux->element, "pad-added",
G_CALLBACK (flvdemux_pad_added), &harnesses);
gst_harness_add_sink_harness (mux, demux);
gst_harness_set_src_caps_str (a_src,
"audio/x-speex, rate=(int)16000, channels=(int)1");
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (a_src,
gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY,
speex_hdr0, sizeof (speex_hdr0), 0, sizeof (speex_hdr0), NULL,
NULL)));
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (a_src,
gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY,
speex_hdr1, sizeof (speex_hdr1), 0, sizeof (speex_hdr1), NULL,
NULL)));
/* Wait a little and make sure no clock was scheduled as this shouldn't happen
* before the caps are set */
g_usleep (40 * 1000);
tclock = gst_harness_get_testclock (mux);
fail_unless (gst_test_clock_get_next_entry_time (tclock) ==
GST_CLOCK_TIME_NONE);
gst_harness_set_src_caps_str (v_src,
"video/x-h264, stream-format=(string)avc, alignment=(string)au, "
"codec_data=(buffer)0142c00cffe1000b6742c00c95a7201e1108d401000468ce3c80");
gst_harness_crank_single_clock_wait (mux);
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (a_src,
gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY,
speex_buf, sizeof (speex_buf), 0, sizeof (speex_buf), NULL,
NULL)));
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (v_src,
gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY,
h264_buf, sizeof (h264_buf), 0, sizeof (h264_buf), NULL, NULL)));
gst_harness_crank_single_clock_wait (mux);
gst_harness_crank_single_clock_wait (mux);
gst_harness_crank_single_clock_wait (mux);
/* push from flvmux to demux */
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 6));
/* verify we got 2x audio and 1x video buffers out of flvdemux */
gst_buffer_unref (gst_harness_pull (a_sink));
gst_buffer_unref (gst_harness_pull (a_sink));
gst_buffer_unref (gst_harness_pull (v_sink));
fail_unless (gst_test_clock_get_next_entry_time (tclock) ==
GST_CLOCK_TIME_NONE);
g_clear_object (&tclock);
gst_harness_teardown (a_src);
gst_harness_teardown (v_src);
gst_harness_teardown (mux);
gst_harness_teardown (a_sink);
gst_harness_teardown (v_sink);
}
GST_END_TEST;
guint8 raw_frame_short[] = {
0x27, 0x00, 0x03, 0x20, 0x64, 0x1c
};
GST_START_TEST (test_video_caps_change_streamable)
{
GstEvent *event;
GstCaps *a_caps1, *v_caps1, *v_caps2, *ret_caps;
GstHarness *mux = gst_harness_new_with_padnames ("flvmux", NULL, "src");
GstHarness *a_src =
gst_harness_new_with_element (mux->element, "audio", NULL);
GstHarness *v_src =
gst_harness_new_with_element (mux->element, "video", NULL);
GstHarness *demux = gst_harness_new_with_padnames ("flvdemux", "sink", NULL);
GstHarness *a_sink =
gst_harness_new_with_element (demux->element, NULL, NULL);
GstHarness *v_sink =
gst_harness_new_with_element (demux->element, NULL, NULL);
DemuxHarnesses harnesses = { a_sink, v_sink };
const GstClockTime base_time = 123456789;
const GstClockTime duration_ms = 20;
const GstClockTime duration = duration_ms * GST_MSECOND;
g_object_set (mux->element, "streamable", TRUE, NULL);
g_signal_connect (demux->element, "pad-added",
G_CALLBACK (flvdemux_pad_added), &harnesses);
gst_harness_add_sink_harness (mux, demux);
a_caps1 =
gst_caps_from_string
("audio/mpeg, mpegversion=(int)4, framed=(boolean)true, stream-format=(string)raw, "
"rate=(int)44100, channels=(int)1, codec_data=(buffer)1208");
v_caps1 = gst_caps_from_string ("video/x-h264, stream-format=(string)avc, "
"codec_data=(buffer)0142c00cffe1000b6742c00c95a7201e1108d401000468ce3c80");
gst_harness_set_src_caps (a_src, gst_caps_ref (a_caps1));
gst_harness_set_src_caps (v_src, gst_caps_ref (v_caps1));
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (a_src,
create_buffer (raw_frame_short, sizeof (raw_frame_short), base_time,
duration)));
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (v_src,
create_buffer (h264_buf, sizeof (h264_buf), base_time, duration)));
gst_harness_crank_single_clock_wait (mux);
/* push from flvmux to demux */
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 6));
/* should accept without the constraint */
while ((event = gst_harness_try_pull_event (v_sink))) {
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
gst_event_parse_caps (event, &ret_caps);
GST_LOG ("v_caps1 %" GST_PTR_FORMAT ", ret caps %" GST_PTR_FORMAT,
v_caps1, ret_caps);
fail_unless (gst_caps_is_equal (v_caps1, ret_caps));
}
gst_event_unref (event);
}
/* caps change */
v_caps2 = gst_caps_from_string ("video/x-h264, stream-format=(string)avc, "
"codec_data=(buffer)0164001fffe1001c6764001facd9405005bb016a02020280000003008000001e478c18cb01000568ebecb22c");
gst_harness_set_src_caps (v_src, gst_caps_ref (v_caps2));
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (v_src,
create_buffer (h264_buf, sizeof (h264_buf), base_time + duration,
duration)));
gst_harness_crank_single_clock_wait (mux);
/* push from flvmux to demux */
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 2));
/* should accept without the constraint */
while ((event = gst_harness_try_pull_event (v_sink))) {
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
gst_event_parse_caps (event, &ret_caps);
GST_LOG ("v_caps2 %" GST_PTR_FORMAT ", ret caps %" GST_PTR_FORMAT,
v_caps2, ret_caps);
fail_unless (gst_caps_is_equal (v_caps2, ret_caps));
}
gst_event_unref (event);
}
/* verify we got 1x audio and 2x video buffers out of flvdemux */
gst_buffer_unref (gst_harness_pull (a_sink));
gst_buffer_unref (gst_harness_pull (v_sink));
gst_buffer_unref (gst_harness_pull (v_sink));
gst_caps_unref (a_caps1);
gst_caps_unref (v_caps1);
gst_caps_unref (v_caps2);
gst_harness_teardown (a_src);
gst_harness_teardown (v_src);
gst_harness_teardown (mux);
gst_harness_teardown (a_sink);
gst_harness_teardown (v_sink);
}
GST_END_TEST;
GST_START_TEST (test_audio_caps_change_streamable)
{
GstEvent *event;
GstCaps *a_caps1, *a_caps2, *v_caps1, *ret_caps;
GstHarness *mux = gst_harness_new_with_padnames ("flvmux", NULL, "src");
GstHarness *a_src =
gst_harness_new_with_element (mux->element, "audio", NULL);
GstHarness *v_src =
gst_harness_new_with_element (mux->element, "video", NULL);
GstHarness *demux = gst_harness_new_with_padnames ("flvdemux", "sink", NULL);
GstHarness *a_sink =
gst_harness_new_with_element (demux->element, NULL, NULL);
GstHarness *v_sink =
gst_harness_new_with_element (demux->element, NULL, NULL);
DemuxHarnesses harnesses = { a_sink, v_sink };
const GstClockTime base_time = 123456789;
const GstClockTime duration_ms = 20;
const GstClockTime duration = duration_ms * GST_MSECOND;
g_object_set (mux->element, "streamable", TRUE, NULL);
g_signal_connect (demux->element, "pad-added",
G_CALLBACK (flvdemux_pad_added), &harnesses);
gst_harness_add_sink_harness (mux, demux);
a_caps1 =
gst_caps_from_string
("audio/mpeg, mpegversion=(int)4, framed=(boolean)true, stream-format=(string)raw, "
"rate=(int)44100, channels=(int)1, codec_data=(buffer)1208");
v_caps1 = gst_caps_from_string ("video/x-h264, stream-format=(string)avc, "
"codec_data=(buffer)0142c00cffe1000b6742c00c95a7201e1108d401000468ce3c80");
gst_harness_set_src_caps (a_src, gst_caps_ref (a_caps1));
gst_harness_set_src_caps (v_src, gst_caps_ref (v_caps1));
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (a_src,
create_buffer (raw_frame_short, sizeof (raw_frame_short), base_time,
duration)));
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (v_src,
create_buffer (h264_buf, sizeof (h264_buf), base_time, duration)));
gst_harness_crank_single_clock_wait (mux);
/* push from flvmux to demux */
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 6));
/* should accept without the constraint */
while ((event = gst_harness_try_pull_event (a_sink))) {
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
gst_event_parse_caps (event, &ret_caps);
GST_LOG ("a_caps1 %" GST_PTR_FORMAT ", ret caps %" GST_PTR_FORMAT,
a_caps1, ret_caps);
fail_unless (gst_caps_is_equal (a_caps1, ret_caps));
}
gst_event_unref (event);
}
/* caps change */
a_caps2 =
gst_caps_from_string
("audio/mpeg, mpegversion=(int)4, framed=(boolean)true, stream-format=(string)raw, "
"rate=(int)22050, channels=(int)1, codec_data=(buffer)1388");
gst_harness_set_src_caps (a_src, gst_caps_ref (a_caps2));
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (a_src,
create_buffer (raw_frame_short, sizeof (raw_frame_short),
base_time + duration, duration)));
gst_harness_crank_single_clock_wait (mux);
/* push from flvmux to demux */
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 2));
/* should accept without the constraint */
while ((event = gst_harness_try_pull_event (a_sink))) {
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
gst_event_parse_caps (event, &ret_caps);
GST_LOG ("a_caps2 %" GST_PTR_FORMAT ", ret caps %" GST_PTR_FORMAT,
a_caps2, ret_caps);
fail_unless (gst_caps_is_equal (a_caps2, ret_caps));
}
gst_event_unref (event);
}
/* verify we got 2x audio and 1x video buffers out of flvdemux */
gst_buffer_unref (gst_harness_pull (a_sink));
gst_buffer_unref (gst_harness_pull (a_sink));
gst_buffer_unref (gst_harness_pull (v_sink));
gst_caps_unref (a_caps1);
gst_caps_unref (a_caps2);
gst_caps_unref (v_caps1);
gst_harness_teardown (a_src);
gst_harness_teardown (v_src);
gst_harness_teardown (mux);
gst_harness_teardown (a_sink);
gst_harness_teardown (v_sink);
}
GST_END_TEST;
GST_START_TEST (test_video_caps_change_streamable_single)
{
GstEvent *event;
GstCaps *v_caps1, *v_caps2, *ret_caps;
GstHarness *mux = gst_harness_new_with_padnames ("flvmux", NULL, "src");
GstHarness *v_src =
gst_harness_new_with_element (mux->element, "video", NULL);
GstHarness *demux = gst_harness_new_with_padnames ("flvdemux", "sink", NULL);
GstHarness *v_sink =
gst_harness_new_with_element (demux->element, NULL, NULL);
DemuxHarnesses harnesses = { NULL, v_sink };
const GstClockTime base_time = 123456789;
const GstClockTime duration_ms = 20;
const GstClockTime duration = duration_ms * GST_MSECOND;
g_object_set (mux->element, "streamable", TRUE, NULL);
g_signal_connect (demux->element, "pad-added",
G_CALLBACK (flvdemux_pad_added), &harnesses);
gst_harness_add_sink_harness (mux, demux);
v_caps1 = gst_caps_from_string ("video/x-h264, stream-format=(string)avc, "
"codec_data=(buffer)0142c00cffe1000b6742c00c95a7201e1108d401000468ce3c80");
gst_harness_set_src_caps (v_src, gst_caps_ref (v_caps1));
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (v_src,
create_buffer (h264_buf, sizeof (h264_buf), base_time, duration)));
/* push from flvmux to demux */
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 4));
/* should accept without the constraint */
while ((event = gst_harness_try_pull_event (v_sink))) {
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
gst_event_parse_caps (event, &ret_caps);
GST_LOG ("v_caps1 %" GST_PTR_FORMAT ", ret caps %" GST_PTR_FORMAT,
v_caps1, ret_caps);
fail_unless (gst_caps_is_equal (v_caps1, ret_caps));
}
gst_event_unref (event);
}
/* caps change */
v_caps2 = gst_caps_from_string ("video/x-h264, stream-format=(string)avc, "
"codec_data=(buffer)0164001fffe1001c6764001facd9405005bb016a02020280000003008000001e478c18cb01000568ebecb22c");
gst_harness_set_src_caps (v_src, gst_caps_ref (v_caps2));
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (v_src,
create_buffer (h264_buf, sizeof (h264_buf), base_time + duration,
duration)));
/* push from flvmux to demux */
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 2));
/* should accept without the constraint */
while ((event = gst_harness_try_pull_event (v_sink))) {
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
gst_event_parse_caps (event, &ret_caps);
GST_LOG ("v_caps2 %" GST_PTR_FORMAT ", ret caps %" GST_PTR_FORMAT,
v_caps2, ret_caps);
fail_unless (gst_caps_is_equal (v_caps2, ret_caps));
}
gst_event_unref (event);
}
/* verify we got 2x video buffers out of flvdemux */
gst_buffer_unref (gst_harness_pull (v_sink));
gst_buffer_unref (gst_harness_pull (v_sink));
gst_caps_unref (v_caps1);
gst_caps_unref (v_caps2);
gst_harness_teardown (v_src);
gst_harness_teardown (mux);
gst_harness_teardown (v_sink);
}
GST_END_TEST;
GST_START_TEST (test_audio_caps_change_streamable_single)
{
GstEvent *event;
GstCaps *a_caps1, *a_caps2, *ret_caps;
GstHarness *mux = gst_harness_new_with_padnames ("flvmux", NULL, "src");
GstHarness *a_src =
gst_harness_new_with_element (mux->element, "audio", NULL);
GstHarness *demux = gst_harness_new_with_padnames ("flvdemux", "sink", NULL);
GstHarness *a_sink =
gst_harness_new_with_element (demux->element, NULL, NULL);
DemuxHarnesses harnesses = { a_sink, NULL };
const GstClockTime base_time = 123456789;
const GstClockTime duration_ms = 20;
const GstClockTime duration = duration_ms * GST_MSECOND;
g_object_set (mux->element, "streamable", TRUE, NULL);
g_signal_connect (demux->element, "pad-added",
G_CALLBACK (flvdemux_pad_added), &harnesses);
gst_harness_add_sink_harness (mux, demux);
a_caps1 =
gst_caps_from_string
("audio/mpeg, mpegversion=(int)4, framed=(boolean)true, stream-format=(string)raw, "
"rate=(int)44100, channels=(int)1, codec_data=(buffer)1208");
gst_harness_set_src_caps (a_src, gst_caps_ref (a_caps1));
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (a_src,
create_buffer (raw_frame_short, sizeof (raw_frame_short), base_time,
duration)));
/* push from flvmux to demux */
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 4));
/* should accept without the constraint */
while ((event = gst_harness_try_pull_event (a_sink))) {
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
gst_event_parse_caps (event, &ret_caps);
GST_LOG ("a_caps1 %" GST_PTR_FORMAT ", ret caps %" GST_PTR_FORMAT,
a_caps1, ret_caps);
fail_unless (gst_caps_is_equal (a_caps1, ret_caps));
}
gst_event_unref (event);
}
/* caps change */
a_caps2 =
gst_caps_from_string
("audio/mpeg, mpegversion=(int)4, framed=(boolean)true, stream-format=(string)raw, "
"rate=(int)22050, channels=(int)1, codec_data=(buffer)1388");
gst_harness_set_src_caps (a_src, gst_caps_ref (a_caps2));
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (a_src,
create_buffer (raw_frame_short, sizeof (raw_frame_short),
base_time + duration, duration)));
/* push from flvmux to demux */
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 2));
/* should accept without the constraint */
while ((event = gst_harness_try_pull_event (a_sink))) {
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
gst_event_parse_caps (event, &ret_caps);
GST_LOG ("a_caps2 %" GST_PTR_FORMAT ", ret caps %" GST_PTR_FORMAT,
a_caps2, ret_caps);
fail_unless (gst_caps_is_equal (a_caps2, ret_caps));
}
gst_event_unref (event);
}
/* verify we got 2x audio out of flvdemux */
gst_buffer_unref (gst_harness_pull (a_sink));
gst_buffer_unref (gst_harness_pull (a_sink));
gst_caps_unref (a_caps1);
gst_caps_unref (a_caps2);
gst_harness_teardown (a_src);
gst_harness_teardown (mux);
gst_harness_teardown (a_sink);
}
GST_END_TEST;
typedef struct
{
guint media_type;
guint64 ts; /* timestamp in ms */
} InputData;
GST_START_TEST (test_incrementing_timestamps)
{
GstPad *audio_sink, *video_sink, *audio_src, *video_src;
GstHarness *h, *audio, *video, *audio_q, *video_q;
guint i;
GstEvent *event;
guint32 prev_pts;
InputData input[] = {
{AUDIO, 155},
{VIDEO, 156},
{VIDEO, 190},
{AUDIO, 176},
{AUDIO, 197},
};
/* setup flvmuxer with queues in front */
h = gst_harness_new_with_padnames ("flvmux", NULL, "src");
audio = gst_harness_new_with_element (h->element, "audio", NULL);
video = gst_harness_new_with_element (h->element, "video", NULL);
audio_q = gst_harness_new ("queue");
video_q = gst_harness_new ("queue");
audio_sink = GST_PAD_PEER (audio->srcpad);
video_sink = GST_PAD_PEER (video->srcpad);
audio_src = GST_PAD_PEER (audio_q->sinkpad);
video_src = GST_PAD_PEER (video_q->sinkpad);
gst_pad_unlink (audio->srcpad, audio_sink);
gst_pad_unlink (video->srcpad, video_sink);
gst_pad_unlink (audio_src, audio_q->sinkpad);
gst_pad_unlink (video_src, video_q->sinkpad);
gst_pad_link (audio_src, audio_sink);
gst_pad_link (video_src, video_sink);
g_object_set (h->element, "streamable", TRUE, NULL);
gst_harness_set_src_caps_str (audio_q,
"audio/mpeg, mpegversion=(int)4, "
"rate=(int)44100, channels=(int)1, "
"stream-format=(string)raw, codec_data=(buffer)1208");
gst_harness_set_src_caps_str (video_q,
"video/x-h264, stream-format=(string)avc, alignment=(string)au, "
"codec_data=(buffer)0142c00dffe1000d6742c00d95a0507c807844235001000468ce3c80");
for (i = 0; i < G_N_ELEMENTS (input); i++) {
InputData *d = &input[i];
GstBuffer *buf = gst_buffer_new ();
GST_BUFFER_DTS (buf) = GST_BUFFER_PTS (buf) = d->ts * GST_MSECOND;
if (d->media_type == AUDIO)
gst_harness_push (audio_q, buf);
else
gst_harness_push (video_q, buf);
}
gst_harness_push_event (audio_q, gst_event_new_eos ());
gst_harness_push_event (video_q, gst_event_new_eos ());
while ((event = gst_harness_pull_event (h)) != NULL) {
GstEventType event_type = GST_EVENT_TYPE (event);
gst_event_unref (event);
if (event_type == GST_EVENT_EOS)
break;
}
/* pull the flv metadata */
gst_buffer_unref (gst_harness_pull (h));
gst_buffer_unref (gst_harness_pull (h));
gst_buffer_unref (gst_harness_pull (h));
/* verify pts in the flvheader is increasing */
prev_pts = 0;
for (i = 0; i < G_N_ELEMENTS (input); i++) {
GstBuffer *buf = gst_harness_pull (h);
GstMapInfo map;
guint32 pts;
fail_unless (buf != NULL);
gst_buffer_map (buf, &map, GST_MAP_READ);
pts = GST_READ_UINT24_BE (map.data + 4);
GST_DEBUG ("media=%u, pts = %u\n", map.data[0], pts);
fail_unless (pts >= prev_pts);
prev_pts = pts;
gst_buffer_unmap (buf, &map);
gst_buffer_unref (buf);
}
/* teardown */
gst_harness_teardown (h);
gst_harness_teardown (audio);
gst_harness_teardown (video);
gst_harness_teardown (audio_q);
gst_harness_teardown (video_q);
}
GST_END_TEST;
GST_START_TEST (test_rollover_timestamps)
{
GstPad *audio_sink, *video_sink, *audio_src, *video_src;
GstHarness *h, *audio, *video, *audio_q, *video_q;
GstEvent *event;
guint i;
guint64 rollover_pts = (guint64) G_MAXUINT32 + 100;
InputData input[] = {
{AUDIO, 0}
,
{VIDEO, 0}
,
{VIDEO, (guint64) G_MAXUINT32 - 100}
,
{AUDIO, (guint64) G_MAXUINT32 - 95}
,
{AUDIO, rollover_pts}
,
};
/* setup flvmuxer with queues in front */
h = gst_harness_new_with_padnames ("flvmux", NULL, "src");
audio = gst_harness_new_with_element (h->element, "audio", NULL);
video = gst_harness_new_with_element (h->element, "video", NULL);
audio_q = gst_harness_new ("queue");
video_q = gst_harness_new ("queue");
audio_sink = GST_PAD_PEER (audio->srcpad);
video_sink = GST_PAD_PEER (video->srcpad);
audio_src = GST_PAD_PEER (audio_q->sinkpad);
video_src = GST_PAD_PEER (video_q->sinkpad);
gst_pad_unlink (audio->srcpad, audio_sink);
gst_pad_unlink (video->srcpad, video_sink);
gst_pad_unlink (audio_src, audio_q->sinkpad);
gst_pad_unlink (video_src, video_q->sinkpad);
gst_pad_link (audio_src, audio_sink);
gst_pad_link (video_src, video_sink);
g_object_set (h->element, "streamable", TRUE, NULL);
gst_harness_set_src_caps_str (audio_q,
"audio/mpeg, mpegversion=(int)4, "
"rate=(int)44100, channels=(int)1, "
"stream-format=(string)raw, codec_data=(buffer)1208");
gst_harness_set_src_caps_str (video_q,
"video/x-h264, stream-format=(string)avc, alignment=(string)au, "
"codec_data=(buffer)0142c00dffe1000d6742c00d95a0507c807844235001000468ce3c80");
for (i = 0; i < G_N_ELEMENTS (input); i++) {
InputData *d = &input[i];
GstBuffer *buf = gst_buffer_new ();
GST_BUFFER_DTS (buf) = GST_BUFFER_PTS (buf) = d->ts * GST_MSECOND;
GST_DEBUG ("Push media=%u, pts=%" G_GUINT64_FORMAT " (%" GST_TIME_FORMAT
")", d->media_type, d->ts, GST_TIME_ARGS (GST_BUFFER_PTS (buf)));
if (d->media_type == AUDIO)
gst_harness_push (audio_q, buf);
else
gst_harness_push (video_q, buf);
}
gst_harness_push_event (audio_q, gst_event_new_eos ());
gst_harness_push_event (video_q, gst_event_new_eos ());
while ((event = gst_harness_pull_event (h)) != NULL) {
GstEventType event_type = GST_EVENT_TYPE (event);
gst_event_unref (event);
if (event_type == GST_EVENT_EOS)
break;
}
/* pull the flv metadata */
gst_buffer_unref (gst_harness_pull (h));
gst_buffer_unref (gst_harness_pull (h));
gst_buffer_unref (gst_harness_pull (h));
gst_buffer_unref (gst_harness_pull (h));
/* verify rollover pts in the flvheader is handled */
for (i = 0; i < G_N_ELEMENTS (input); i++) {
GstBuffer *buf = gst_harness_pull (h);
GstMapInfo map;
guint32 pts, pts_ext;
fail_unless (buf != NULL);
gst_buffer_map (buf, &map, GST_MAP_READ);
pts = GST_READ_UINT24_BE (map.data + 4);
pts_ext = GST_READ_UINT8 (map.data + 7);
pts |= pts_ext << 24;
GST_DEBUG ("media=%u, pts=%u (%" GST_TIME_FORMAT ")",
map.data[0], pts, GST_TIME_ARGS (pts * GST_MSECOND));
fail_unless (pts == (guint32) input[i].ts);
gst_buffer_unmap (buf, &map);
gst_buffer_unref (buf);
}
/* teardown */
gst_harness_teardown (h);
gst_harness_teardown (audio);
gst_harness_teardown (video);
gst_harness_teardown (audio_q);
gst_harness_teardown (video_q);
}
GST_END_TEST;
static Suite *
flvmux_suite (void)
{
Suite *s = suite_create ("flvmux");
TCase *tc_chain = tcase_create ("general");
gint loop = 16;
suite_add_tcase (s, tc_chain);
#ifdef HAVE_VALGRIND
if (RUNNING_ON_VALGRIND) {
loop = 1;
}
#endif
tcase_add_loop_test (tc_chain, test_index_writing, 0, loop);
tcase_add_test (tc_chain, test_speex_streamable);
tcase_add_test (tc_chain, test_increasing_timestamp_when_pts_none);
tcase_add_test (tc_chain, test_video_caps_late);
tcase_add_test (tc_chain, test_audio_caps_change_streamable);
tcase_add_test (tc_chain, test_video_caps_change_streamable);
tcase_add_test (tc_chain, test_audio_caps_change_streamable_single);
tcase_add_test (tc_chain, test_video_caps_change_streamable_single);
tcase_add_test (tc_chain, test_incrementing_timestamps);
tcase_add_test (tc_chain, test_rollover_timestamps);
return s;
}
GST_CHECK_MAIN (flvmux)