mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-25 11:11:08 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/tests/check/elements/qtmux.c
Guillaume Desmottes 0f719af307 tests: matroskamux, qtmux: don't add codec_data buffers to template caps 
streamheader and codec_data buffers fields are only meant to be
in the negotiated caps, not the template caps.

Fixes false-positive leaks of those buffers detected by the leaks
tracer, as template caps are static, and we decided to not include
code in gstreamer core to handle this unusual case of template caps
having buffers in them.

https://bugzilla.gnome.org/show_bug.cgi?id=768762
2017-02-21 15:47:16 +00:00

1628 lines
51 KiB
C
Raw Blame History

/* GStreamer
*
* unit test for qtmux
*
* Copyright (C) <2008> Mark Nauwelaerts <mnauw@users.sf.net>
*
* 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_UNISTD_H
#include <unistd.h>
#endif
#include <glib/gstdio.h>
#include <gst/check/gstcheck.h>
#include <gst/pbutils/encoding-profile.h>
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
static GstPad *mysrcpad, *mysinkpad;
#define VIDEO_RAW_CAPS_STRING "video/x-raw"
#define AUDIO_CAPS_STRING "audio/mpeg, " \
"mpegversion = (int) 1, " \
"layer = (int) 3, " \
"channels = (int) 2, " \
"rate = (int) 48000"
#define AUDIO_AAC_TMPL_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 shouldn't be in the template caps, only in the actual caps */
#define AUDIO_AAC_CAPS_STRING AUDIO_AAC_TMPL_CAPS_STRING \
", codec_data=(buffer)1208"
#define VIDEO_CAPS_STRING "video/mpeg, " \
"mpegversion = (int) 4, " \
"systemstream = (boolean) false, " \
"width = (int) 384, " \
"height = (int) 288, " \
"framerate = (fraction) 25/1"
#define VIDEO_TMPL_CAPS_H264_STRING "video/x-h264, " \
"width=(int)320, " \
"height=(int)240, " \
"framerate=(fraction)30/1, " \
"pixel-aspect-ratio=(fraction)1/1, " \
"stream-format=(string)avc, " \
"alignment=(string)au, " \
"level=(string)2, " \
"profile=(string)high"
/* codec_data shouldn't be in the template caps, only in the actual caps */
#define VIDEO_CAPS_H264_STRING VIDEO_TMPL_CAPS_H264_STRING \
", codec_data=(buffer)01640014ffe1001867640014a" \
"cd94141fb0110000003001773594000f14299600" \
"1000568ebecb22c"
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/quicktime"));
static GstStaticPadTemplate srcvideotemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (VIDEO_CAPS_STRING));
static GstStaticPadTemplate srcvideoh264template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (VIDEO_TMPL_CAPS_H264_STRING));
static GstStaticPadTemplate srcvideorawtemplate =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (VIDEO_RAW_CAPS_STRING));
static GstStaticPadTemplate srcaudiotemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
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_TMPL_CAPS_STRING));
/* setup and teardown needs some special handling for muxer */
static GstPad *
setup_src_pad (GstElement * element,
GstStaticPadTemplate * template, const gchar * sinkname)
{
GstPad *srcpad, *sinkpad;
GST_DEBUG_OBJECT (element, "setting up sending pad");
/* sending pad */
srcpad = gst_pad_new_from_static_template (template, "src");
fail_if (srcpad == NULL, "Could not create a srcpad");
ASSERT_OBJECT_REFCOUNT (srcpad, "srcpad", 1);
if (!(sinkpad = gst_element_get_static_pad (element, sinkname)))
sinkpad = gst_element_get_request_pad (element, sinkname);
fail_if (sinkpad == NULL, "Could not get sink pad from %s",
GST_ELEMENT_NAME (element));
/* references are owned by: 1) us, 2) qtmux, 3) collect pads */
ASSERT_OBJECT_REFCOUNT (sinkpad, "sinkpad", 3);
fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK,
"Could not link source and %s sink pads", GST_ELEMENT_NAME (element));
gst_object_unref (sinkpad); /* because we got it higher up */
/* references are owned by: 1) qtmux, 2) collect pads */
ASSERT_OBJECT_REFCOUNT (sinkpad, "sinkpad", 2);
return srcpad;
}
static void
teardown_src_pad (GstPad * srcpad)
{
GstPad *sinkpad;
/* clean up floating src pad */
sinkpad = gst_pad_get_peer (srcpad);
fail_if (sinkpad == NULL);
/* pad refs held by 1) qtmux 2) collectpads and 3) us (through _get_peer) */
ASSERT_OBJECT_REFCOUNT (sinkpad, "sinkpad", 3);
gst_pad_unlink (srcpad, sinkpad);
/* after unlinking, pad refs still held by
* 1) qtmux and 2) collectpads and 3) us (through _get_peer) */
ASSERT_OBJECT_REFCOUNT (sinkpad, "sinkpad", 3);
gst_object_unref (sinkpad);
/* one more ref is held by element itself */
/* pad refs held by creator */
ASSERT_OBJECT_REFCOUNT (srcpad, "srcpad", 1);
gst_object_unref (srcpad);
}
gboolean downstream_is_seekable;
static gboolean
qtmux_sinkpad_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
gboolean ret = FALSE;
if (GST_QUERY_TYPE (query) == GST_QUERY_SEEKING) {
gst_query_set_seeking (query, GST_FORMAT_BYTES, downstream_is_seekable, 0,
-1);
ret = TRUE;
}
return ret;
}
static GstElement *
setup_qtmux (GstStaticPadTemplate * srctemplate, const gchar * sinkname,
gboolean seekable)
{
GstElement *qtmux;
GST_DEBUG ("setup_qtmux");
qtmux = gst_check_setup_element ("qtmux");
mysrcpad = setup_src_pad (qtmux, srctemplate, sinkname);
mysinkpad = gst_check_setup_sink_pad (qtmux, &sinktemplate);
downstream_is_seekable = seekable;
gst_pad_set_query_function (mysinkpad, qtmux_sinkpad_query);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
return qtmux;
}
static void
cleanup_qtmux (GstElement * qtmux, const gchar * sinkname)
{
GST_DEBUG ("cleanup_qtmux");
gst_element_set_state (qtmux, GST_STATE_NULL);
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
teardown_src_pad (mysrcpad);
gst_check_teardown_sink_pad (qtmux);
gst_check_teardown_element (qtmux);
}
static void
check_qtmux_pad (GstStaticPadTemplate * srctemplate, const gchar * sinkname,
guint32 dts_method)
{
GstElement *qtmux;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
int num_buffers;
int i;
guint8 data0[12] = "\000\000\000\024ftypqt ";
guint8 data1[16] = "\000\000\000\010free\000\000\000\000mdat";
guint8 data2[4] = "moov";
GstSegment segment;
qtmux = setup_qtmux (srctemplate, sinkname, TRUE);
g_object_set (qtmux, "dts-method", dts_method, NULL);
fail_unless (gst_element_set_state (qtmux,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
gst_pad_push_event (mysrcpad, gst_event_new_stream_start ("test"));
caps = gst_pad_get_pad_template_caps (mysrcpad);
gst_pad_set_caps (mysrcpad, caps);
gst_caps_unref (caps);
/* ensure segment (format) properly setup */
gst_segment_init (&segment, GST_FORMAT_TIME);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
inbuffer = gst_buffer_new_and_alloc (1);
gst_buffer_memset (inbuffer, 0, 0, 1);
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
GST_BUFFER_DURATION (inbuffer) = 40 * GST_MSECOND;
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
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);
num_buffers = g_list_length (buffers);
/* at least expect ftyp, mdat header, buffer chunk and moov */
fail_unless (num_buffers >= 4);
/* clean up first to clear any pending refs in sticky caps */
cleanup_qtmux (qtmux, sinkname);
for (i = 0; i < num_buffers; ++i) {
outbuffer = GST_BUFFER (buffers->data);
fail_if (outbuffer == NULL);
buffers = g_list_remove (buffers, outbuffer);
switch (i) {
case 0:
{
/* ftyp header */
fail_unless (gst_buffer_get_size (outbuffer) >= 20);
fail_unless (gst_buffer_memcmp (outbuffer, 0, data0,
sizeof (data0)) == 0);
fail_unless (gst_buffer_memcmp (outbuffer, 16, data0 + 8, 4) == 0);
break;
}
case 1: /* mdat header */
fail_unless (gst_buffer_get_size (outbuffer) == 16);
fail_unless (gst_buffer_memcmp (outbuffer, 0, data1, sizeof (data1))
== 0);
break;
case 2: /* buffer we put in */
fail_unless (gst_buffer_get_size (outbuffer) == 1);
break;
case 3: /* moov */
fail_unless (gst_buffer_get_size (outbuffer) > 8);
fail_unless (gst_buffer_memcmp (outbuffer, 4, data2,
sizeof (data2)) == 0);
break;
default:
break;
}
ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
gst_buffer_unref (outbuffer);
outbuffer = NULL;
}
g_list_free (buffers);
buffers = NULL;
}
static void
check_qtmux_pad_fragmented (GstStaticPadTemplate * srctemplate,
const gchar * sinkname, guint32 dts_method, gboolean streamable)
{
GstElement *qtmux;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
int num_buffers;
int i;
guint8 data0[12] = "\000\000\000\024ftypqt ";
guint8 data1[4] = "mdat";
guint8 data2[4] = "moov";
guint8 data3[4] = "moof";
guint8 data4[4] = "mfra";
GstSegment segment;
qtmux = setup_qtmux (srctemplate, sinkname, !streamable);
g_object_set (qtmux, "dts-method", dts_method, NULL);
g_object_set (qtmux, "fragment-duration", 2000, NULL);
g_object_set (qtmux, "streamable", streamable, NULL);
fail_unless (gst_element_set_state (qtmux,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
gst_pad_push_event (mysrcpad, gst_event_new_stream_start ("test"));
caps = gst_pad_get_pad_template_caps (mysrcpad);
gst_pad_set_caps (mysrcpad, caps);
gst_caps_unref (caps);
/* ensure segment (format) properly setup */
gst_segment_init (&segment, GST_FORMAT_TIME);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
inbuffer = gst_buffer_new_and_alloc (1);
gst_buffer_memset (inbuffer, 0, 0, 1);
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
GST_BUFFER_DURATION (inbuffer) = 40 * GST_MSECOND;
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* send eos to have all written */
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()) == TRUE);
num_buffers = g_list_length (buffers);
/* at least expect ftyp, moov, moof, mdat header, buffer chunk
* and optionally mfra */
fail_unless (num_buffers >= 5);
/* clean up first to clear any pending refs in sticky caps */
cleanup_qtmux (qtmux, sinkname);
for (i = 0; i < num_buffers; ++i) {
outbuffer = GST_BUFFER (buffers->data);
fail_if (outbuffer == NULL);
buffers = g_list_remove (buffers, outbuffer);
switch (i) {
case 0:
{
/* ftyp header */
fail_unless (gst_buffer_get_size (outbuffer) >= 20);
fail_unless (gst_buffer_memcmp (outbuffer, 0, data0,
sizeof (data0)) == 0);
fail_unless (gst_buffer_memcmp (outbuffer, 16, data0 + 8, 4) == 0);
break;
}
case 1: /* moov */
fail_unless (gst_buffer_get_size (outbuffer) > 8);
fail_unless (gst_buffer_memcmp (outbuffer, 4, data2,
sizeof (data2)) == 0);
break;
case 2: /* moof */
fail_unless (gst_buffer_get_size (outbuffer) > 8);
fail_unless (gst_buffer_memcmp (outbuffer, 4, data3,
sizeof (data3)) == 0);
break;
case 3: /* mdat header */
fail_unless (gst_buffer_get_size (outbuffer) == 8);
fail_unless (gst_buffer_memcmp (outbuffer, 4, data1,
sizeof (data1)) == 0);
break;
case 4: /* buffer we put in */
fail_unless (gst_buffer_get_size (outbuffer) == 1);
break;
case 5: /* mfra */
fail_unless (gst_buffer_get_size (outbuffer) > 8);
fail_unless (gst_buffer_memcmp (outbuffer, 4, data4,
sizeof (data4)) == 0);
break;
default:
break;
}
ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
gst_buffer_unref (outbuffer);
outbuffer = NULL;
}
g_list_free (buffers);
buffers = NULL;
}
/* dts-method dd */
GST_START_TEST (test_video_pad_dd)
{
check_qtmux_pad (&srcvideotemplate, "video_%u", 0);
}
GST_END_TEST;
GST_START_TEST (test_audio_pad_dd)
{
check_qtmux_pad (&srcaudiotemplate, "audio_%u", 0);
}
GST_END_TEST;
GST_START_TEST (test_video_pad_frag_dd)
{
check_qtmux_pad_fragmented (&srcvideotemplate, "video_%u", 0, FALSE);
}
GST_END_TEST;
GST_START_TEST (test_audio_pad_frag_dd)
{
check_qtmux_pad_fragmented (&srcaudiotemplate, "audio_%u", 0, FALSE);
}
GST_END_TEST;
GST_START_TEST (test_video_pad_frag_dd_streamable)
{
check_qtmux_pad_fragmented (&srcvideotemplate, "video_%u", 0, TRUE);
}
GST_END_TEST;
GST_START_TEST (test_audio_pad_frag_dd_streamable)
{
check_qtmux_pad_fragmented (&srcaudiotemplate, "audio_%u", 0, TRUE);
}
GST_END_TEST;
/* dts-method reorder */
GST_START_TEST (test_video_pad_reorder)
{
check_qtmux_pad (&srcvideotemplate, "video_%u", 1);
}
GST_END_TEST;
GST_START_TEST (test_audio_pad_reorder)
{
check_qtmux_pad (&srcaudiotemplate, "audio_%u", 1);
}
GST_END_TEST;
GST_START_TEST (test_video_pad_frag_reorder)
{
check_qtmux_pad_fragmented (&srcvideotemplate, "video_%u", 1, FALSE);
}
GST_END_TEST;
GST_START_TEST (test_audio_pad_frag_reorder)
{
check_qtmux_pad_fragmented (&srcaudiotemplate, "audio_%u", 1, FALSE);
}
GST_END_TEST;
GST_START_TEST (test_video_pad_frag_reorder_streamable)
{
check_qtmux_pad_fragmented (&srcvideotemplate, "video_%u", 1, TRUE);
}
GST_END_TEST;
GST_START_TEST (test_audio_pad_frag_reorder_streamable)
{
check_qtmux_pad_fragmented (&srcaudiotemplate, "audio_%u", 1, TRUE);
}
GST_END_TEST;
/* dts-method asc */
GST_START_TEST (test_video_pad_asc)
{
check_qtmux_pad (&srcvideotemplate, "video_%u", 2);
}
GST_END_TEST;
GST_START_TEST (test_audio_pad_asc)
{
check_qtmux_pad (&srcaudiotemplate, "audio_%u", 2);
}
GST_END_TEST;
GST_START_TEST (test_video_pad_frag_asc)
{
check_qtmux_pad_fragmented (&srcvideotemplate, "video_%u", 2, FALSE);
}
GST_END_TEST;
GST_START_TEST (test_audio_pad_frag_asc)
{
check_qtmux_pad_fragmented (&srcaudiotemplate, "audio_%u", 2, FALSE);
}
GST_END_TEST;
GST_START_TEST (test_video_pad_frag_asc_streamable)
{
check_qtmux_pad_fragmented (&srcvideotemplate, "video_%u", 2, TRUE);
}
GST_END_TEST;
GST_START_TEST (test_audio_pad_frag_asc_streamable)
{
check_qtmux_pad_fragmented (&srcaudiotemplate, "audio_%u", 2, TRUE);
}
GST_END_TEST;
GST_START_TEST (test_reuse)
{
GstElement *qtmux = setup_qtmux (&srcvideotemplate, "video_%u", TRUE);
GstBuffer *inbuffer;
GstCaps *caps;
GstSegment segment;
gst_element_set_state (qtmux, GST_STATE_PLAYING);
gst_element_set_state (qtmux, GST_STATE_NULL);
gst_element_set_state (qtmux, GST_STATE_PLAYING);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
gst_pad_push_event (mysrcpad, gst_event_new_stream_start ("test"));
caps = gst_pad_get_pad_template_caps (mysrcpad);
gst_pad_set_caps (mysrcpad, caps);
gst_caps_unref (caps);
/* ensure segment (format) properly setup */
gst_segment_init (&segment, GST_FORMAT_TIME);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
inbuffer = gst_buffer_new_and_alloc (1);
fail_unless (inbuffer != NULL);
gst_buffer_memset (inbuffer, 0, 0, 1);
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
GST_BUFFER_DURATION (inbuffer) = 40 * GST_MSECOND;
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* send eos to have all written */
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()) == TRUE);
cleanup_qtmux (qtmux, "video_%u");
gst_check_drop_buffers ();
}
GST_END_TEST;
static GstEncodingContainerProfile *
create_qtmux_profile (const gchar * variant)
{
GstEncodingContainerProfile *cprof;
GstCaps *caps;
if (variant == NULL) {
caps = gst_caps_new_empty_simple ("video/quicktime");
} else {
caps = gst_caps_new_simple ("video/quicktime",
"variant", G_TYPE_STRING, variant, NULL);
}
cprof = gst_encoding_container_profile_new ("Name", "blah", caps, NULL);
gst_caps_unref (caps);
caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, "S16BE",
"channels", G_TYPE_INT, 2, "rate", G_TYPE_INT, 44100, NULL);
gst_encoding_container_profile_add_profile (cprof,
GST_ENCODING_PROFILE (gst_encoding_audio_profile_new (caps, NULL, NULL,
1)));
gst_caps_unref (caps);
return cprof;
}
GST_START_TEST (test_encodebin_qtmux)
{
GstEncodingContainerProfile *cprof;
GstElement *enc;
GstPad *pad;
enc = gst_element_factory_make ("encodebin", NULL);
if (enc == NULL)
return;
/* Make sure encodebin finds a muxer for a profile with a variant field .. */
cprof = create_qtmux_profile ("apple");
g_object_set (enc, "profile", cprof, NULL);
gst_encoding_profile_unref (cprof);
/* should have created a pad after setting the profile */
pad = gst_element_get_static_pad (enc, "audio_0");
fail_unless (pad != NULL);
gst_object_unref (pad);
gst_object_unref (enc);
/* ... and for a profile without a variant field */
enc = gst_element_factory_make ("encodebin", NULL);
cprof = create_qtmux_profile (NULL);
g_object_set (enc, "profile", cprof, NULL);
gst_encoding_profile_unref (cprof);
/* should have created a pad after setting the profile */
pad = gst_element_get_static_pad (enc, "audio_0");
fail_unless (pad != NULL);
gst_object_unref (pad);
gst_object_unref (enc);
}
GST_END_TEST;
/* Fake mp3 encoder for test */
typedef GstElement TestMp3Enc;
typedef GstElementClass TestMp3EncClass;
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg, mpegversion=1, layer=[1,3]")
);
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw")
);
static GType test_mp3_enc_get_type (void);
static void test_input_push_segment_start (gpointer user_data,
GstClockTime start);
G_DEFINE_TYPE (TestMp3Enc, test_mp3_enc, GST_TYPE_ELEMENT);
static void
test_mp3_enc_class_init (TestMp3EncClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_static_pad_template (element_class, &sink_template);
gst_element_class_add_static_pad_template (element_class, &src_template);
gst_element_class_set_metadata (element_class, "MPEG1 Audio Encoder",
"Codec/Encoder/Audio", "Pretends to encode mp3", "Foo Bar <foo@bar.com>");
}
static void
test_mp3_enc_init (TestMp3Enc * mp3enc)
{
GstPad *pad;
pad = gst_pad_new_from_static_template (&sink_template, "sink");
gst_element_add_pad (mp3enc, pad);
pad = gst_pad_new_from_static_template (&src_template, "src");
gst_element_add_pad (mp3enc, pad);
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "testmp3enc", GST_RANK_NONE,
test_mp3_enc_get_type ());
}
static GstEncodingContainerProfile *
create_mp4mux_profile (void)
{
GstEncodingContainerProfile *cprof;
GstCaps *caps;
caps = gst_caps_new_simple ("video/quicktime",
"variant", G_TYPE_STRING, "iso", NULL);
cprof = gst_encoding_container_profile_new ("Name", "blah", caps, NULL);
gst_caps_unref (caps);
caps = gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 1,
"layer", G_TYPE_INT, 3, "channels", G_TYPE_INT, 2, "rate", G_TYPE_INT,
44100, NULL);
gst_encoding_container_profile_add_profile (cprof,
GST_ENCODING_PROFILE (gst_encoding_audio_profile_new (caps, NULL, NULL,
1)));
gst_caps_unref (caps);
return cprof;
}
GST_START_TEST (test_encodebin_mp4mux)
{
GstEncodingContainerProfile *cprof;
GstPluginFeature *feature;
GstElement *enc, *mux;
GstPad *pad;
/* need a fake mp3 encoder because mp4 only accepts encoded formats */
gst_plugin_register_static (GST_VERSION_MAJOR, GST_VERSION_MINOR,
"fakemp3enc", "fakemp3enc", plugin_init, VERSION, "LGPL",
"gst-plugins-good", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
feature = gst_registry_find_feature (gst_registry_get (), "testmp3enc",
GST_TYPE_ELEMENT_FACTORY);
gst_plugin_feature_set_rank (feature, GST_RANK_PRIMARY + 100);
enc = gst_element_factory_make ("encodebin", NULL);
if (enc == NULL)
return;
/* Make sure encodebin finds mp4mux even though qtmux outputs a superset */
cprof = create_mp4mux_profile ();
g_object_set (enc, "profile", cprof, NULL);
gst_encoding_profile_unref (cprof);
/* should have created a pad after setting the profile */
pad = gst_element_get_static_pad (enc, "audio_0");
fail_unless (pad != NULL);
gst_object_unref (pad);
mux = gst_bin_get_by_interface (GST_BIN (enc), GST_TYPE_TAG_SETTER);
fail_unless (mux != NULL);
{
GstElementFactory *f = gst_element_get_factory (mux);
/* make sure we got mp4mux for variant=iso */
GST_INFO ("muxer: %s", G_OBJECT_TYPE_NAME (mux));
fail_unless_equals_string (GST_OBJECT_NAME (f), "mp4mux");
}
gst_object_unref (mux);
gst_object_unref (enc);
gst_plugin_feature_set_rank (feature, GST_RANK_NONE);
gst_object_unref (feature);
}
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_empty ();
}
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_unref (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;
}
static void
test_average_bitrate_custom (const gchar * elementname,
GstStaticPadTemplate * tmpl, const gchar * caps_str,
const gchar * sinkpadname)
{
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;
GstSegment segment;
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 (elementname);
filesink = gst_element_factory_make ("filesink", NULL);
g_object_set (filesink, "location", location, NULL);
gst_element_link (qtmux, filesink);
mysrcpad = setup_src_pad (qtmux, tmpl, sinkpadname);
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");
gst_pad_push_event (mysrcpad, gst_event_new_stream_start ("test"));
caps = gst_caps_from_string (caps_str);
gst_pad_set_caps (mysrcpad, caps);
gst_caps_unref (caps);
/* ensure segment (format) properly setup */
gst_segment_init (&segment, GST_FORMAT_TIME);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
for (i = 0; i < 3; i++) {
inbuffer = gst_buffer_new_and_alloc (bytes[i]);
gst_buffer_memset (inbuffer, 0, 0, bytes[i]);
GST_BUFFER_TIMESTAMP (inbuffer) = total_duration;
GST_BUFFER_DURATION (inbuffer) = (GstClockTime) durations[i];
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
total_bytes += gst_buffer_get_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_check_drop_buffers ();
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_tag_list_unref (taglist);
}
/* delete file */
g_unlink (location);
g_free (location);
}
GST_START_TEST (test_average_bitrate)
{
test_average_bitrate_custom ("mp4mux", &srcaudioaactemplate,
AUDIO_AAC_CAPS_STRING, "audio_%u");
test_average_bitrate_custom ("mp4mux", &srcvideoh264template,
VIDEO_CAPS_H264_STRING, "video_%u");
test_average_bitrate_custom ("qtmux", &srcaudioaactemplate,
AUDIO_AAC_CAPS_STRING, "audio_%u");
test_average_bitrate_custom ("qtmux", &srcvideoh264template,
VIDEO_CAPS_H264_STRING, "video_%u");
}
GST_END_TEST;
struct TestInputData
{
GstPad *srcpad;
GstSegment segment;
GList *input;
GThread *thread;
/* When comparing ts, the input will be subtracted from this */
gint64 ts_offset;
/* Due to DTS, the segment start might be shifted so this list
* is used to vefity each received segments */
GList *expected_segment_start;
GstClockTime expected_gap_ts;
GstClockTime expected_gap_duration;
gboolean gap_received;
GstPad *sinkpad;
GList *output_iter;
};
static void
test_input_data_init (struct TestInputData *data)
{
data->ts_offset = 0;
data->expected_segment_start = NULL;
data->expected_gap_ts = 0;
data->expected_gap_duration = 0;
data->gap_received = FALSE;
data->srcpad = NULL;
data->sinkpad = NULL;
data->input = NULL;
data->thread = NULL;
test_input_push_segment_start (data, 0);
}
static void
test_input_data_clean (struct TestInputData *data)
{
g_list_free_full (data->input, (GDestroyNotify) gst_mini_object_unref);
if (data->sinkpad) {
gst_pad_set_active (data->sinkpad, FALSE);
gst_object_unref (data->sinkpad);
}
gst_pad_set_active (data->srcpad, FALSE);
teardown_src_pad (data->srcpad);
}
static gpointer
test_input_push_data (gpointer user_data)
{
struct TestInputData *data = user_data;
GList *iter;
GstFlowReturn flow;
for (iter = data->input; iter; iter = g_list_next (iter)) {
if (GST_IS_BUFFER (iter->data)) {
GST_INFO ("Pushing buffer %" GST_PTR_FORMAT " on pad: %s:%s", iter->data,
GST_DEBUG_PAD_NAME (data->srcpad));
flow =
gst_pad_push (data->srcpad,
gst_buffer_ref ((GstBuffer *) iter->data));
fail_unless (flow == GST_FLOW_OK);
} else {
GST_INFO_OBJECT (data->srcpad, "Pushing event: %"
GST_PTR_FORMAT, iter->data);
fail_unless (gst_pad_push_event (data->srcpad,
gst_event_ref ((GstEvent *) iter->data)) == TRUE);
}
}
return NULL;
}
static void
test_input_push_segment_start (gpointer user_data, GstClockTime start)
{
struct TestInputData *data = user_data;
GstClockTime *start_data = g_malloc (sizeof (GstClockTime));
*start_data = start;
data->expected_segment_start = g_list_append (data->expected_segment_start,
start_data);
}
static GstClockTime
test_input_pop_segment_start (gpointer user_data)
{
struct TestInputData *data = user_data;
GstClockTime start = GST_CLOCK_TIME_NONE;
GstClockTime *start_data;
if (data->expected_segment_start) {
start_data = data->expected_segment_start->data;
data->expected_segment_start =
g_list_delete_link (data->expected_segment_start,
data->expected_segment_start);
start = *start_data;
g_free (start_data);
}
return start;
}
static GstBuffer *
create_buffer (GstClockTime pts, GstClockTime dts, GstClockTime duration,
guint bytes)
{
GstBuffer *buf;
guint8 *data;
data = g_malloc0 (bytes);
buf = gst_buffer_new_wrapped (data, bytes);
GST_BUFFER_PTS (buf) = pts;
GST_BUFFER_DTS (buf) = dts;
GST_BUFFER_DURATION (buf) = duration;
return buf;
}
static GstFlowReturn
_test_sink_pad_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
struct TestInputData *test_data = g_object_get_qdata (G_OBJECT (pad),
g_quark_from_static_string ("test-mux-pad"));
GstBuffer *expected_buffer;
fail_unless (test_data->output_iter);
fail_unless (GST_IS_BUFFER (test_data->output_iter->data));
expected_buffer = test_data->output_iter->data;
fail_unless (GST_BUFFER_PTS (buffer) ==
(GST_BUFFER_PTS_IS_VALID (expected_buffer) ?
GST_BUFFER_PTS (expected_buffer) -
test_data->ts_offset : GST_BUFFER_PTS (expected_buffer)));
fail_unless (GST_BUFFER_DTS (buffer) ==
(GST_BUFFER_DTS_IS_VALID (expected_buffer) ?
GST_BUFFER_DTS (expected_buffer) -
test_data->ts_offset : GST_BUFFER_DTS (buffer)));
fail_unless (GST_BUFFER_DURATION (buffer) ==
GST_BUFFER_DURATION (expected_buffer));
test_data->output_iter = g_list_next (test_data->output_iter);
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
static void
compare_event (GstEvent * event, GstEvent * expected)
{
fail_unless (GST_EVENT_TYPE (event) == GST_EVENT_TYPE (expected));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:{
GstCaps *caps, *expected_caps;
gst_event_parse_caps (event, &caps);
gst_event_parse_caps (expected, &expected_caps);
fail_unless (gst_caps_can_intersect (caps, expected_caps));
}
break;
default:
break;
}
}
static gboolean
_test_sink_pad_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
struct TestInputData *test_data = g_object_get_qdata (G_OBJECT (pad),
g_quark_from_static_string ("test-mux-pad"));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_STREAM_START:
case GST_EVENT_CAPS:
case GST_EVENT_EOS:
fail_unless (test_data->output_iter);
fail_unless (GST_IS_EVENT (test_data->output_iter->data));
compare_event (event, test_data->output_iter->data);
test_data->output_iter = g_list_next (test_data->output_iter);
break;
case GST_EVENT_SEGMENT:{
const GstSegment *segment;
fail_unless (test_data->output_iter);
fail_unless (GST_IS_EVENT (test_data->output_iter->data));
gst_event_parse_segment (event, &segment);
fail_unless (segment->start == test_input_pop_segment_start (test_data));
test_data->output_iter = g_list_next (test_data->output_iter);
break;
}
case GST_EVENT_GAP:{
GstClockTime timestamp;
GstClockTime duration;
gst_event_parse_gap (event, &timestamp, &duration);
fail_unless (timestamp == test_data->expected_gap_ts);
fail_unless (duration == test_data->expected_gap_duration);
test_data->gap_received = TRUE;
break;
}
case GST_EVENT_TAG:
/* ignore this event */
break;
default:
GST_ERROR_OBJECT (pad, "Unexpected event: %" GST_PTR_FORMAT, event);
fail ("Unexpected event received %s", GST_EVENT_TYPE_NAME (event));
break;
}
gst_event_unref (event);
return TRUE;
}
static void
_test_pad_added_cb (GstElement * element, GstPad * pad, gpointer udata)
{
GstCaps *caps;
struct TestInputData **inputs = udata;
gint i = -1;
const gchar *name;
const gchar *strname;
caps = gst_pad_get_current_caps (pad);
strname = gst_structure_get_name (gst_caps_get_structure (caps, 0));
if (g_str_has_prefix (strname, "video/")) {
i = 0; /* video is 0, audio is 1 */
name = "videosink";
} else {
i = 1;
name = "audiosink";
}
gst_caps_unref (caps);
fail_unless (i != -1);
fail_unless (inputs[i]->sinkpad == NULL);
inputs[i]->sinkpad = gst_pad_new (name, GST_PAD_SINK);
inputs[i]->output_iter = inputs[i]->input;
g_object_set_qdata (G_OBJECT (inputs[i]->sinkpad),
g_quark_from_static_string ("test-mux-pad"), inputs[i]);
gst_pad_set_chain_function (inputs[i]->sinkpad, _test_sink_pad_chain);
gst_pad_set_event_function (inputs[i]->sinkpad, _test_sink_pad_event);
gst_pad_set_active (inputs[i]->sinkpad, TRUE);
fail_unless (gst_pad_link (pad, inputs[i]->sinkpad) == GST_PAD_LINK_OK);
}
static void
check_output (const gchar * location, struct TestInputData *input1,
struct TestInputData *input2)
{
GstElement *filesrc;
GstElement *demux;
struct TestInputData *inputs[2] = { input1, input2 };
filesrc = gst_element_factory_make ("filesrc", NULL);
demux = gst_element_factory_make ("qtdemux", NULL);
fail_unless (gst_element_link (filesrc, demux));
g_object_set (filesrc, "location", location, NULL);
g_signal_connect (demux, "pad-added", (GCallback) _test_pad_added_cb, inputs);
fail_unless (gst_element_set_state (demux,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
fail_unless (gst_element_set_state (filesrc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
/* FIXME use a main loop */
g_usleep (2 * G_USEC_PER_SEC);
fail_unless (gst_element_set_state (demux,
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS);
fail_unless (gst_element_set_state (filesrc,
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS);
gst_object_unref (filesrc);
gst_object_unref (demux);
}
/* Muxes a file with qtmux using the inputs provided and
* then verifies that the generated file corresponds to the
* data in the inputs */
static void
run_muxing_test (struct TestInputData *input1, struct TestInputData *input2)
{
gchar *location;
GstElement *qtmux;
GstElement *filesink;
location = g_strdup_printf ("%s/%s-%d", g_get_tmp_dir (), "qtmuxtest",
g_random_int ());
qtmux = gst_check_setup_element ("qtmux");
filesink = gst_element_factory_make ("filesink", NULL);
g_object_set (filesink, "location", location, NULL);
gst_element_link (qtmux, filesink);
input1->srcpad = setup_src_pad (qtmux, &srcvideorawtemplate, "video_%u");
fail_unless (input1->srcpad != NULL);
gst_pad_set_active (input1->srcpad, TRUE);
input2->srcpad = setup_src_pad (qtmux, &srcaudioaactemplate, "audio_%u");
fail_unless (input2->srcpad != NULL);
gst_pad_set_active (input2->srcpad, 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");
input1->thread =
g_thread_new ("test-push-data-1", test_input_push_data, input1);
input2->thread =
g_thread_new ("test-push-data-2", test_input_push_data, input2);
/* FIXME set a mainloop and wait for EOS */
g_thread_join (input1->thread);
g_thread_join (input2->thread);
input1->thread = NULL;
input2->thread = NULL;
gst_element_set_state (qtmux, GST_STATE_NULL);
gst_element_set_state (filesink, GST_STATE_NULL);
check_output (location, input1, input2);
gst_object_unref (filesink);
test_input_data_clean (input1);
test_input_data_clean (input2);
gst_check_teardown_element (qtmux);
/* delete file */
g_unlink (location);
g_free (location);
}
GST_START_TEST (test_muxing)
{
struct TestInputData input1, input2;
GstCaps *caps;
test_input_data_init (&input1);
test_input_data_init (&input2);
/* Create the inputs, after calling the run below, all this data is
* transfered to it and we have no need to clean up */
input1.input = NULL;
input1.input =
g_list_append (input1.input, gst_event_new_stream_start ("test-1"));
caps = gst_caps_from_string
("video/x-raw, width=(int)800, height=(int)600, "
"framerate=(fraction)1/1, format=(string)RGB");
input1.input = g_list_append (input1.input, gst_event_new_caps (caps));
gst_caps_unref (caps);
gst_segment_init (&input1.segment, GST_FORMAT_TIME);
input1.input =
g_list_append (input1.input, gst_event_new_segment (&input1.segment));
input1.input =
g_list_append (input1.input, create_buffer (0, GST_CLOCK_TIME_NONE,
GST_SECOND, 800 * 600 * 3));
input1.input =
g_list_append (input1.input, create_buffer (1 * GST_SECOND,
GST_CLOCK_TIME_NONE, GST_SECOND, 800 * 600 * 3));
input1.input =
g_list_append (input1.input, create_buffer (2 * GST_SECOND,
GST_CLOCK_TIME_NONE, GST_SECOND, 800 * 600 * 3));
input1.input = g_list_append (input1.input, gst_event_new_eos ());
input2.input = NULL;
input2.input =
g_list_append (input2.input, gst_event_new_stream_start ("test-2"));
caps = gst_caps_from_string (AUDIO_AAC_CAPS_STRING);
input2.input = g_list_append (input2.input, gst_event_new_caps (caps));
gst_caps_unref (caps);
gst_segment_init (&input2.segment, GST_FORMAT_TIME);
input2.input =
g_list_append (input2.input, gst_event_new_segment (&input2.segment));
input2.input =
g_list_append (input2.input, create_buffer (0, 0, GST_SECOND, 4096));
input2.input =
g_list_append (input2.input, create_buffer (1 * GST_SECOND,
1 * GST_SECOND, GST_SECOND, 4096));
input2.input =
g_list_append (input2.input, create_buffer (2 * GST_SECOND,
2 * GST_SECOND, GST_SECOND, 4096));
input2.input = g_list_append (input2.input, gst_event_new_eos ());
run_muxing_test (&input1, &input2);
}
GST_END_TEST;
GST_START_TEST (test_muxing_non_zero_segment)
{
struct TestInputData input1, input2;
GstCaps *caps;
test_input_data_init (&input1);
test_input_data_init (&input2);
/* Create the inputs, after calling the run below, all this data is
* transfered to it and we have no need to clean up */
input1.input = NULL;
input1.input =
g_list_append (input1.input, gst_event_new_stream_start ("test-1"));
caps = gst_caps_from_string
("video/x-raw, width=(int)800, height=(int)600, "
"framerate=(fraction)1/1, format=(string)RGB");
input1.input = g_list_append (input1.input, gst_event_new_caps (caps));
gst_caps_unref (caps);
gst_segment_init (&input1.segment, GST_FORMAT_TIME);
input1.segment.start = 10 * GST_SECOND;
input1.input =
g_list_append (input1.input, gst_event_new_segment (&input1.segment));
input1.input =
g_list_append (input1.input, create_buffer (10 * GST_SECOND,
GST_CLOCK_TIME_NONE, GST_SECOND, 800 * 600 * 3));
input1.input =
g_list_append (input1.input, create_buffer (11 * GST_SECOND,
GST_CLOCK_TIME_NONE, GST_SECOND, 800 * 600 * 3));
input1.input =
g_list_append (input1.input, create_buffer (12 * GST_SECOND,
GST_CLOCK_TIME_NONE, GST_SECOND, 800 * 600 * 3));
input1.input = g_list_append (input1.input, gst_event_new_eos ());
input1.ts_offset = GST_SECOND * 10;
input2.input = NULL;
input2.input =
g_list_append (input2.input, gst_event_new_stream_start ("test-2"));
caps = gst_caps_from_string (AUDIO_AAC_CAPS_STRING);
input2.input = g_list_append (input2.input, gst_event_new_caps (caps));
gst_caps_unref (caps);
gst_segment_init (&input2.segment, GST_FORMAT_TIME);
input2.segment.start = 10 * GST_SECOND;
input2.input =
g_list_append (input2.input, gst_event_new_segment (&input2.segment));
input2.input =
g_list_append (input2.input, create_buffer (10 * GST_SECOND,
10 * GST_SECOND, GST_SECOND, 4096));
input2.input =
g_list_append (input2.input, create_buffer (11 * GST_SECOND,
11 * GST_SECOND, GST_SECOND, 4096));
input2.input =
g_list_append (input2.input, create_buffer (12 * GST_SECOND,
12 * GST_SECOND, GST_SECOND, 4096));
input2.input = g_list_append (input2.input, gst_event_new_eos ());
input2.ts_offset = GST_SECOND * 10;
run_muxing_test (&input1, &input2);
}
GST_END_TEST;
GST_START_TEST (test_muxing_non_zero_segment_different)
{
struct TestInputData input1, input2;
GstCaps *caps;
test_input_data_init (&input1);
test_input_data_init (&input2);
/* Create the inputs, after calling the run below, all this data is
* transfered to it and we have no need to clean up */
input1.input = NULL;
input1.input =
g_list_append (input1.input, gst_event_new_stream_start ("test-1"));
caps = gst_caps_from_string
("video/x-raw, width=(int)800, height=(int)600, "
"framerate=(fraction)1/1, format=(string)RGB");
input1.input = g_list_append (input1.input, gst_event_new_caps (caps));
gst_caps_unref (caps);
gst_segment_init (&input1.segment, GST_FORMAT_TIME);
input1.segment.start = 5 * GST_SECOND;
input1.input =
g_list_append (input1.input, gst_event_new_segment (&input1.segment));
input1.input =
g_list_append (input1.input, create_buffer (5 * GST_SECOND,
GST_CLOCK_TIME_NONE, GST_SECOND, 800 * 600 * 3));
input1.input =
g_list_append (input1.input, create_buffer (6 * GST_SECOND,
GST_CLOCK_TIME_NONE, GST_SECOND, 800 * 600 * 3));
input1.input =
g_list_append (input1.input, create_buffer (7 * GST_SECOND,
GST_CLOCK_TIME_NONE, GST_SECOND, 800 * 600 * 3));
input1.input = g_list_append (input1.input, gst_event_new_eos ());
input1.ts_offset = GST_SECOND * 5;
input2.input = NULL;
input2.input =
g_list_append (input2.input, gst_event_new_stream_start ("test-2"));
caps = gst_caps_from_string (AUDIO_AAC_CAPS_STRING);
input2.input = g_list_append (input2.input, gst_event_new_caps (caps));
gst_caps_unref (caps);
gst_segment_init (&input2.segment, GST_FORMAT_TIME);
input2.segment.start = 10 * GST_SECOND;
input2.input =
g_list_append (input2.input, gst_event_new_segment (&input2.segment));
input2.input =
g_list_append (input2.input, create_buffer (10 * GST_SECOND,
10 * GST_SECOND, GST_SECOND, 4096));
input2.input =
g_list_append (input2.input, create_buffer (11 * GST_SECOND,
11 * GST_SECOND, GST_SECOND, 4096));
input2.input =
g_list_append (input2.input, create_buffer (12 * GST_SECOND,
12 * GST_SECOND, GST_SECOND, 4096));
input2.input = g_list_append (input2.input, gst_event_new_eos ());
input2.ts_offset = GST_SECOND * 10;
run_muxing_test (&input1, &input2);
}
GST_END_TEST;
GST_START_TEST (test_muxing_dts_outside_segment)
{
struct TestInputData input1, input2;
GstCaps *caps;
test_input_data_init (&input1);
test_input_data_init (&input2);
/* Create the inputs, after calling the run below, all this data is
* transfered to it and we have no need to clean up */
input1.input = NULL;
input1.input =
g_list_append (input1.input, gst_event_new_stream_start ("test-1"));
caps = gst_caps_from_string
("video/x-h264, width=(int)800, height=(int)600, "
"framerate=(fraction)1/1, stream-format=(string)avc, codec_data=(buffer)0000,"
" alignment=(string)au, level=(int)2, profile=(string)high");
input1.input = g_list_append (input1.input, gst_event_new_caps (caps));
gst_caps_unref (caps);
gst_segment_init (&input1.segment, GST_FORMAT_TIME);
input1.segment.start = 1 * GST_SECOND;
input1.input =
g_list_append (input1.input, gst_event_new_segment (&input1.segment));
input1.input =
g_list_append (input1.input, create_buffer (1 * GST_SECOND,
0, GST_SECOND, 4096));
input1.input =
g_list_append (input1.input, create_buffer (2 * GST_SECOND,
1 * GST_SECOND, GST_SECOND, 4096));
input1.input =
g_list_append (input1.input, create_buffer (3 * GST_SECOND,
2 * GST_SECOND, GST_SECOND, 4096));
input1.input = g_list_append (input1.input, gst_event_new_eos ());
/* First DTS is 0, first PTS is 1s. The segment start being 1, this means
* running time -1s and 0. So the output segment should start from 1s to keep
* the same running time */
test_input_pop_segment_start (&input1);
test_input_push_segment_start (&input1, GST_SECOND);
input2.input = NULL;
input2.input =
g_list_append (input2.input, gst_event_new_stream_start ("test-2"));
caps = gst_caps_from_string (AUDIO_AAC_CAPS_STRING);
input2.input = g_list_append (input2.input, gst_event_new_caps (caps));
gst_caps_unref (caps);
gst_segment_init (&input2.segment, GST_FORMAT_TIME);
input2.input =
g_list_append (input2.input, gst_event_new_segment (&input2.segment));
input2.input =
g_list_append (input2.input, create_buffer (0, 0, GST_SECOND,
44100 * 4 * 2));
input2.input =
g_list_append (input2.input, create_buffer (GST_SECOND, GST_SECOND,
GST_SECOND, 44100 * 4 * 2));
input2.input =
g_list_append (input2.input, create_buffer (2 * GST_SECOND,
2 * GST_SECOND, GST_SECOND, 44100 * 4 * 2));
input2.input = g_list_append (input2.input, gst_event_new_eos ());
run_muxing_test (&input1, &input2);
}
GST_END_TEST;
GST_START_TEST (test_muxing_initial_gap)
{
struct TestInputData input1, input2;
GstCaps *caps;
test_input_data_init (&input1);
test_input_data_init (&input2);
/* Create the inputs, after calling the run below, all this data is
* transfered to it and we have no need to clean up */
input1.input = NULL;
input1.input =
g_list_append (input1.input, gst_event_new_stream_start ("test-1"));
caps = gst_caps_from_string
("video/x-h264, width=(int)800, height=(int)600, "
"framerate=(fraction)1/1, stream-format=(string)avc, codec_data=(buffer)0000,"
" alignment=(string)au, level=(int)2, profile=(string)high");
input1.input = g_list_append (input1.input, gst_event_new_caps (caps));
gst_caps_unref (caps);
gst_segment_init (&input1.segment, GST_FORMAT_TIME);
input1.input =
g_list_append (input1.input, gst_event_new_segment (&input1.segment));
/* Duplicate the segment to please the harness */
input1.input =
g_list_append (input1.input, gst_event_new_segment (&input1.segment));
input1.input =
g_list_append (input1.input, create_buffer (1 * GST_SECOND,
0, GST_SECOND, 4096));
input1.input =
g_list_append (input1.input, create_buffer (2 * GST_SECOND,
1 * GST_SECOND, GST_SECOND, 4096));
input1.input =
g_list_append (input1.input, create_buffer (3 * GST_SECOND,
2 * GST_SECOND, GST_SECOND, 4096));
input1.input = g_list_append (input1.input, gst_event_new_eos ());
/* We expect a 1s gap at the start */
input1.expected_gap_duration = GST_SECOND;
/* There will be two segments, first is 0, so leave it there, second should
* match the first CTTS (PTS - DTS) */
test_input_push_segment_start (&input1, GST_SECOND);
input2.input = NULL;
input2.input =
g_list_append (input2.input, gst_event_new_stream_start ("test-2"));
caps = gst_caps_from_string (AUDIO_AAC_CAPS_STRING);
input2.input = g_list_append (input2.input, gst_event_new_caps (caps));
gst_caps_unref (caps);
gst_segment_init (&input2.segment, GST_FORMAT_TIME);
input2.input =
g_list_append (input2.input, gst_event_new_segment (&input2.segment));
input2.input =
g_list_append (input2.input, create_buffer (0, 0, GST_SECOND,
44100 * 4 * 2));
input2.input =
g_list_append (input2.input, create_buffer (GST_SECOND, GST_SECOND,
GST_SECOND, 44100 * 4 * 2));
input2.input =
g_list_append (input2.input, create_buffer (2 * GST_SECOND,
2 * GST_SECOND, GST_SECOND, 44100 * 4 * 2));
input2.input = g_list_append (input2.input, gst_event_new_eos ());
run_muxing_test (&input1, &input2);
fail_unless (input1.gap_received);
}
GST_END_TEST;
static Suite *
qtmux_suite (void)
{
Suite *s = suite_create ("qtmux");
TCase *tc_chain = tcase_create ("general");
/* avoid glib warnings when setting deprecated dts-method property */
g_setenv ("G_ENABLE_DIAGNOSTIC", "0", TRUE);
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_video_pad_dd);
tcase_add_test (tc_chain, test_audio_pad_dd);
tcase_add_test (tc_chain, test_video_pad_frag_dd);
tcase_add_test (tc_chain, test_audio_pad_frag_dd);
tcase_add_test (tc_chain, test_video_pad_frag_dd_streamable);
tcase_add_test (tc_chain, test_audio_pad_frag_dd_streamable);
tcase_add_test (tc_chain, test_video_pad_reorder);
tcase_add_test (tc_chain, test_audio_pad_reorder);
tcase_add_test (tc_chain, test_video_pad_frag_reorder);
tcase_add_test (tc_chain, test_audio_pad_frag_reorder);
tcase_add_test (tc_chain, test_video_pad_frag_reorder_streamable);
tcase_add_test (tc_chain, test_audio_pad_frag_reorder_streamable);
tcase_add_test (tc_chain, test_video_pad_asc);
tcase_add_test (tc_chain, test_audio_pad_asc);
tcase_add_test (tc_chain, test_video_pad_frag_asc);
tcase_add_test (tc_chain, test_audio_pad_frag_asc);
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_average_bitrate);
tcase_add_test (tc_chain, test_reuse);
tcase_add_test (tc_chain, test_encodebin_qtmux);
tcase_add_test (tc_chain, test_encodebin_mp4mux);
tcase_add_test (tc_chain, test_muxing);
tcase_add_test (tc_chain, test_muxing_non_zero_segment);
tcase_add_test (tc_chain, test_muxing_non_zero_segment_different);
tcase_add_test (tc_chain, test_muxing_dts_outside_segment);
tcase_add_test (tc_chain, test_muxing_initial_gap);
return s;
}
GST_CHECK_MAIN (qtmux)
Reference in a new issue View git blame Copy permalink