tests: qtmux: add tests for bitrate average calculation

Adds tests to make sure qtmux/mp4mux sets average bitrate
correctly
This commit is contained in:
Thiago Santos 2011-09-30 10:55:53 -03:00
parent 7a143ea94f
commit e58b0466ec

View file

@ -41,6 +41,17 @@ static GstPad *mysrcpad, *mysinkpad;
"layer = (int) 3, " \ "layer = (int) 3, " \
"channels = (int) 2, " \ "channels = (int) 2, " \
"rate = (int) 48000" "rate = (int) 48000"
#define AUDIO_AAC_CAPS_STRING "audio/mpeg, " \
"mpegversion=(int)4, " \
"channels=(int)1, " \
"rate=(int)44100, " \
"stream-format=(string)raw, " \
"level=(string)2, " \
"base-profile=(string)lc, " \
"profile=(string)lc, " \
"codec_data=(buffer)1208"
#define VIDEO_CAPS_STRING "video/mpeg, " \ #define VIDEO_CAPS_STRING "video/mpeg, " \
"mpegversion = (int) 4, " \ "mpegversion = (int) 4, " \
"systemstream = (boolean) false, " \ "systemstream = (boolean) false, " \
@ -62,6 +73,11 @@ static GstStaticPadTemplate srcaudiotemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_ALWAYS, GST_PAD_ALWAYS,
GST_STATIC_CAPS (AUDIO_CAPS_STRING)); GST_STATIC_CAPS (AUDIO_CAPS_STRING));
static GstStaticPadTemplate srcaudioaactemplate =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (AUDIO_AAC_CAPS_STRING));
/* setup and teardown needs some special handling for muxer */ /* setup and teardown needs some special handling for muxer */
static GstPad * static GstPad *
@ -684,6 +700,130 @@ GST_START_TEST (test_encodebin_mp4mux)
GST_END_TEST; GST_END_TEST;
static gboolean
extract_tags (const gchar * location, GstTagList ** taglist)
{
gboolean ret = TRUE;
GstElement *src;
GstBus *bus;
GstElement *pipeline =
gst_parse_launch ("filesrc name=src ! qtdemux ! fakesink", NULL);
src = gst_bin_get_by_name (GST_BIN (pipeline), "src");
g_object_set (src, "location", location, NULL);
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
fail_unless (gst_element_set_state (pipeline, GST_STATE_PLAYING)
!= GST_STATE_CHANGE_FAILURE);
if (*taglist == NULL) {
*taglist = gst_tag_list_new ();
}
while (1) {
GstMessage *msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
GST_MESSAGE_TAG | GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_EOS) {
gst_message_unref (msg);
break;
} else if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR) {
ret = FALSE;
gst_message_unref (msg);
break;
} else if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_TAG) {
GstTagList *tags;
gst_message_parse_tag (msg, &tags);
gst_tag_list_insert (*taglist, tags, GST_TAG_MERGE_REPLACE);
gst_tag_list_free (tags);
}
gst_message_unref (msg);
}
gst_object_unref (bus);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (src);
gst_object_unref (pipeline);
return ret;
}
GST_START_TEST (test_average_bitrate)
{
gchar *location;
GstElement *qtmux;
GstElement *filesink;
GstBuffer *inbuffer;
GstCaps *caps;
int i;
gint bytes[] = { 16, 22, 12 };
gint64 durations[] = { GST_SECOND * 3, GST_SECOND * 5, GST_SECOND * 2 };
gint64 total_bytes = 0;
GstClockTime total_duration = 0;
location = g_strdup_printf ("%s/%s-%d", g_get_tmp_dir (), "qtmuxtest",
g_random_int ());
GST_INFO ("Using location %s for bitrate test", location);
qtmux = gst_check_setup_element ("mp4mux");
filesink = gst_element_factory_make ("filesink", NULL);
g_object_set (filesink, "location", location, NULL);
gst_element_link (qtmux, filesink);
mysrcpad = setup_src_pad (qtmux, &srcaudioaactemplate, NULL, "audio_%d");
fail_unless (mysrcpad != NULL);
gst_pad_set_active (mysrcpad, TRUE);
fail_unless (gst_element_set_state (filesink,
GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE,
"could not set filesink to playing");
fail_unless (gst_element_set_state (qtmux,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
for (i = 0; i < 3; i++) {
inbuffer = gst_buffer_new_and_alloc (bytes[i]);
caps = gst_caps_copy (gst_pad_get_pad_template_caps (mysrcpad));
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
GST_BUFFER_TIMESTAMP (inbuffer) = total_duration;
GST_BUFFER_DURATION (inbuffer) = (GstClockTime) durations[i];
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
total_bytes += GST_BUFFER_SIZE (inbuffer);
total_duration += GST_BUFFER_DURATION (inbuffer);
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
}
/* send eos to have moov written */
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()) == TRUE);
gst_element_set_state (qtmux, GST_STATE_NULL);
gst_element_set_state (filesink, GST_STATE_NULL);
gst_pad_set_active (mysrcpad, FALSE);
teardown_src_pad (mysrcpad);
gst_object_unref (filesink);
gst_check_teardown_element (qtmux);
/* check the bitrate tag */
{
GstTagList *taglist = NULL;
guint bitrate = 0;
guint expected;
fail_unless (extract_tags (location, &taglist));
fail_unless (gst_tag_list_get_uint (taglist, GST_TAG_BITRATE, &bitrate));
expected =
(guint) gst_util_uint64_scale_round ((guint64) total_bytes,
(guint64) 8 * GST_SECOND, (guint64) total_duration);
fail_unless (bitrate == expected);
}
}
GST_END_TEST;
static Suite * static Suite *
qtmux_suite (void) qtmux_suite (void)
{ {
@ -712,6 +852,8 @@ qtmux_suite (void)
tcase_add_test (tc_chain, test_video_pad_frag_asc_streamable); tcase_add_test (tc_chain, test_video_pad_frag_asc_streamable);
tcase_add_test (tc_chain, test_audio_pad_frag_asc_streamable); tcase_add_test (tc_chain, test_audio_pad_frag_asc_streamable);
tcase_add_test (tc_chain, test_average_bitrate);
tcase_add_test (tc_chain, test_reuse); tcase_add_test (tc_chain, test_reuse);
tcase_add_test (tc_chain, test_encodebin_qtmux); tcase_add_test (tc_chain, test_encodebin_qtmux);
tcase_add_test (tc_chain, test_encodebin_mp4mux); tcase_add_test (tc_chain, test_encodebin_mp4mux);