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coral / mtk-gst-plugins-good-debian / 607cc70968a2c1d2272195c1a52a41b16f1b8fda / . / tests / check / elements / qtmux.c
blob: a029c524e0732ffd7906e903ed0d6d1a7b778635 [file] [log] [blame]
/* 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_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, " \
"mpegversion = (int) 4, " \
"systemstream = (boolean) false, " \
"width = (int) 384, " \
"height = (int) 288, " \
"framerate = (fraction) 25/1"
#define VIDEO_CAPS_H264_STRING "video/x-h264, " \
"width=(int)320, " \
"height=(int)240, " \
"framerate=(fraction)30/1, " \
"pixel-aspect-ratio=(fraction)1/1, " \
"codec_data=(buffer)01640014ffe1001867640014a" \
"cd94141fb0110000003001773594000f14299600" \
"1000568ebecb22c, " \
"stream-format=(string)avc, " \
"alignment=(string)au, " \
"level=(string)2, " \
"profile=(string)high"
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_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_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 * 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_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)));
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_%u");
test_average_bitrate_custom ("mp4mux", &srcvideoh264template, "video_%u");
test_average_bitrate_custom ("qtmux", &srcaudioaactemplate, "audio_%u");
test_average_bitrate_custom ("qtmux", &srcvideoh264template, "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)