gstreamer/subprojects/gst-plugins-base/tests/check/elements/audiorate.c

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/* GStreamer unit tests for audiorate
*
* Copyright (C) 2006 Tim-Philipp Müller <tim centricular 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
#include <gst/check/gstcheck.h>
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#include <gst/audio/audio.h>
#include <gst/app/gstappsrc.h>
/* helper element to insert additional buffers overlapping with previous ones */
static gdouble injector_inject_probability = 0.0;
typedef GstElement TestInjector;
typedef GstElementClass TestInjectorClass;
GType test_injector_get_type (void);
G_DEFINE_TYPE (TestInjector, test_injector, GST_TYPE_ELEMENT);
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#define FORMATS "{ "GST_AUDIO_NE(F32)", S8, S16LE, S16BE, " \
"U16LE, U16NE, S32LE, S32BE, U32LE, U32BE }"
#define INJECTOR_CAPS \
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"audio/x-raw, " \
"format = (string) "FORMATS", " \
"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ]"
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (INJECTOR_CAPS));
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (INJECTOR_CAPS));
static void
test_injector_class_init (TestInjectorClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_static_pad_template (element_class, &src_template);
gst_element_class_add_static_pad_template (element_class, &sink_template);
}
static GstFlowReturn
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test_injector_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstFlowReturn ret;
GstPad *srcpad;
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srcpad = gst_element_get_static_pad (GST_ELEMENT (parent), "src");
/* since we're increasing timestamp/offsets, push this one first */
GST_LOG (" passing buffer [t=%" GST_TIME_FORMAT "-%" GST_TIME_FORMAT
"], offset=%" G_GINT64_FORMAT ", offset_end=%" G_GINT64_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf)),
GST_BUFFER_OFFSET (buf), GST_BUFFER_OFFSET_END (buf));
gst_buffer_ref (buf);
ret = gst_pad_push (srcpad, buf);
if (g_random_double () < injector_inject_probability) {
GstBuffer *ibuf;
ibuf = gst_buffer_copy (buf);
if (GST_BUFFER_OFFSET_IS_VALID (buf) &&
GST_BUFFER_OFFSET_END_IS_VALID (buf)) {
guint64 delta;
delta = GST_BUFFER_OFFSET_END (buf) - GST_BUFFER_OFFSET (buf);
GST_BUFFER_OFFSET (ibuf) += delta / 4;
GST_BUFFER_OFFSET_END (ibuf) += delta / 4;
} else {
GST_BUFFER_OFFSET (ibuf) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_OFFSET_END (ibuf) = GST_BUFFER_OFFSET_NONE;
}
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf) &&
GST_BUFFER_DURATION_IS_VALID (buf)) {
GstClockTime delta;
delta = GST_BUFFER_DURATION (buf);
GST_BUFFER_TIMESTAMP (ibuf) += delta / 4;
} else {
GST_BUFFER_TIMESTAMP (ibuf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (ibuf) = GST_CLOCK_TIME_NONE;
}
if (GST_BUFFER_TIMESTAMP_IS_VALID (ibuf) ||
GST_BUFFER_OFFSET_IS_VALID (ibuf)) {
GST_LOG ("injecting buffer [t=%" GST_TIME_FORMAT "-%" GST_TIME_FORMAT
"], offset=%" G_GINT64_FORMAT ", offset_end=%" G_GINT64_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (ibuf)),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (ibuf) +
GST_BUFFER_DURATION (ibuf)), GST_BUFFER_OFFSET (ibuf),
GST_BUFFER_OFFSET_END (ibuf));
if (gst_pad_push (srcpad, ibuf) != GST_FLOW_OK) {
/* ignore return value */
}
} else {
GST_WARNING ("couldn't inject buffer, no incoming timestamps or offsets");
gst_buffer_unref (ibuf);
}
}
gst_buffer_unref (buf);
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gst_object_unref (srcpad);
return ret;
}
static void
test_injector_init (TestInjector * injector)
{
GstPad *pad;
pad = gst_pad_new_from_static_template (&sink_template, "sink");
gst_pad_set_chain_function (pad, test_injector_chain);
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GST_PAD_SET_PROXY_CAPS (pad);
gst_element_add_pad (GST_ELEMENT (injector), pad);
pad = gst_pad_new_from_static_template (&src_template, "src");
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GST_PAD_SET_PROXY_CAPS (pad);
gst_element_add_pad (GST_ELEMENT (injector), pad);
}
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static GstPadProbeReturn
probe_cb (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
{
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GstBuffer *buf = GST_PAD_PROBE_INFO_BUFFER (info);
gdouble *drop_probability = user_data;
if (g_random_double () < *drop_probability) {
GST_LOG ("dropping buffer [t=%" GST_TIME_FORMAT "-%" GST_TIME_FORMAT "], "
"offset=%" G_GINT64_FORMAT ", offset_end=%" G_GINT64_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf)),
GST_BUFFER_OFFSET (buf), GST_BUFFER_OFFSET_END (buf));
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return GST_PAD_PROBE_DROP; /* drop buffer */
}
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return GST_PAD_PROBE_OK; /* don't drop buffer */
}
static void
got_buf (GstElement * fakesink, GstBuffer * buf, GstPad * pad, GList ** p_bufs)
{
*p_bufs = g_list_append (*p_bufs, gst_buffer_ref (buf));
}
static void
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do_perfect_stream_test (guint rate, const gchar * format,
gdouble drop_probability, gdouble inject_probability)
{
GstElement *pipe, *src, *conv, *filter, *injector, *audiorate, *sink;
GstMessage *msg;
GstCaps *caps;
GstPad *srcpad;
GList *l, *bufs = NULL;
GstClockTime next_time = GST_CLOCK_TIME_NONE;
guint64 next_offset = GST_BUFFER_OFFSET_NONE;
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GstAudioFormat fmt;
const GstAudioFormatInfo *finfo;
gint width;
fmt = gst_audio_format_from_string (format);
fail_unless (fmt != GST_AUDIO_FORMAT_UNKNOWN);
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finfo = gst_audio_format_get_info (fmt);
width = GST_AUDIO_FORMAT_INFO_WIDTH (finfo);
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caps = gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT,
rate, "format", G_TYPE_STRING, format, NULL);
GST_INFO ("-------- drop=%.0f%% caps = %" GST_PTR_FORMAT " ---------- ",
drop_probability * 100.0, caps);
g_assert (drop_probability >= 0.0 && drop_probability <= 1.0);
g_assert (inject_probability >= 0.0 && inject_probability <= 1.0);
pipe = gst_pipeline_new ("pipeline");
fail_unless (pipe != NULL);
src = gst_element_factory_make ("audiotestsrc", "audiotestsrc");
fail_unless (src != NULL);
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g_object_set (src, "num-buffers", 10, NULL);
conv = gst_element_factory_make ("audioconvert", "audioconvert");
fail_unless (conv != NULL);
filter = gst_element_factory_make ("capsfilter", "capsfilter");
fail_unless (filter != NULL);
g_object_set (filter, "caps", caps, NULL);
injector_inject_probability = inject_probability;
injector = GST_ELEMENT (g_object_new (test_injector_get_type (), NULL));
Don't use bad gst_element_get_pad(). Original commit message from CVS: * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind): * gst/playback/decodetest.c: (new_decoded_pad_cb): * gst/playback/gstdecodebin.c: (gst_decode_bin_init), (try_to_link_1), (elem_is_dynamic), (close_link), (type_found), (cleanup_decodebin): * gst/playback/gstdecodebin2.c: (gst_decode_bin_init), (connect_element), (gst_decode_group_control_demuxer_pad): * gst/playback/gstplaybasebin.c: (queue_remove_probe), (queue_out_of_data), (gen_preroll_element), (preroll_unlinked), (mute_group_type): * gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked), (gst_play_bin_set_property), (handoff), (gen_video_element), (gen_text_element), (gen_audio_element), (gen_vis_element), (remove_sinks), (add_sink), (setup_sinks): * gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb): * gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink), (gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked), (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin), (gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute), (gen_video_chain), (gen_text_chain), (gen_audio_chain), (gen_vis_chain), (gst_play_sink_reconfigure), (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc), (gst_play_sink_request_pad): * gst/playback/gsturidecodebin.c: (type_found), (setup_source): * gst/playback/test.c: (gen_video_element), (gen_audio_element), (cb_newpad): * gst/playback/test6.c: (new_decoded_pad_cb): * tests/check/elements/audioconvert.c: (GST_START_TEST): * tests/check/elements/audiorate.c: (test_injector_chain), (do_perfect_stream_test): * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST): * tests/check/elements/gdpdepay.c: (GST_START_TEST): * tests/check/elements/gnomevfssink.c: * tests/check/elements/textoverlay.c: (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2): * tests/check/elements/videotestsrc.c: (GST_START_TEST): * tests/check/libs/cddabasesrc.c: (GST_START_TEST): * tests/check/pipelines/oggmux.c: (test_pipeline): * tests/check/pipelines/streamheader.c: (GST_START_TEST): * tests/check/pipelines/theoraenc.c: (GST_START_TEST): * tests/check/pipelines/vorbisenc.c: (GST_START_TEST): * tests/examples/seek/scrubby.c: (make_wav_pipeline): * tests/examples/seek/seek.c: (make_mod_pipeline), (make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline), (make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline), (make_theora_pipeline), (make_vorbis_theora_pipeline), (make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline), (make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline), (update_fill), (msg_buffering): Don't use bad gst_element_get_pad().
2008-05-21 16:36:50 +00:00
srcpad = gst_element_get_static_pad (injector, "src");
fail_unless (srcpad != NULL);
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gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_BUFFER, probe_cb,
&drop_probability, NULL);
gst_object_unref (srcpad);
audiorate = gst_element_factory_make ("audiorate", "audiorate");
fail_unless (audiorate != NULL);
sink = gst_element_factory_make ("fakesink", "fakesink");
fail_unless (sink != NULL);
g_object_set (sink, "signal-handoffs", TRUE, NULL);
g_signal_connect (sink, "handoff", G_CALLBACK (got_buf), &bufs);
gst_bin_add_many (GST_BIN (pipe), src, conv, filter, injector, audiorate,
sink, NULL);
gst_element_link_many (src, conv, filter, injector, audiorate, sink, NULL);
fail_unless_equals_int (gst_element_set_state (pipe, GST_STATE_PLAYING),
GST_STATE_CHANGE_ASYNC);
fail_unless_equals_int (gst_element_get_state (pipe, NULL, NULL, -1),
GST_STATE_CHANGE_SUCCESS);
msg = gst_bus_poll (GST_ELEMENT_BUS (pipe),
GST_MESSAGE_EOS | GST_MESSAGE_ERROR, -1);
fail_unless_equals_string (GST_MESSAGE_TYPE_NAME (msg), "eos");
for (l = bufs; l != NULL; l = l->next) {
GstBuffer *buf = GST_BUFFER (l->data);
guint num_samples;
fail_unless (GST_BUFFER_TIMESTAMP_IS_VALID (buf));
fail_unless (GST_BUFFER_DURATION_IS_VALID (buf));
fail_unless (GST_BUFFER_OFFSET_IS_VALID (buf));
fail_unless (GST_BUFFER_OFFSET_END_IS_VALID (buf));
GST_LOG ("buffer: ts=%" GST_TIME_FORMAT ", end_ts=%" GST_TIME_FORMAT
" off=%" G_GINT64_FORMAT ", end_off=%" G_GINT64_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf)),
GST_BUFFER_OFFSET (buf), GST_BUFFER_OFFSET_END (buf));
if (GST_CLOCK_TIME_IS_VALID (next_time)) {
fail_unless_equals_uint64 (next_time, GST_BUFFER_TIMESTAMP (buf));
}
if (next_offset != GST_BUFFER_OFFSET_NONE) {
fail_unless_equals_uint64 (next_offset, GST_BUFFER_OFFSET (buf));
}
/* check buffer size for sanity */
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fail_unless_equals_int (gst_buffer_get_size (buf) % (width / 8), 0);
/* check there is actually as much data as there should be */
num_samples = GST_BUFFER_OFFSET_END (buf) - GST_BUFFER_OFFSET (buf);
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fail_unless_equals_int (gst_buffer_get_size (buf),
num_samples * (width / 8));
next_time = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
next_offset = GST_BUFFER_OFFSET_END (buf);
}
gst_message_unref (msg);
gst_element_set_state (pipe, GST_STATE_NULL);
gst_object_unref (pipe);
g_list_foreach (bufs, (GFunc) gst_mini_object_unref, NULL);
g_list_free (bufs);
gst_caps_unref (caps);
}
static const guint rates[] = { 8000, 11025, 16000, 22050, 32000, 44100,
48000, 3333, 33333, 66666, 9999
};
GST_START_TEST (test_perfect_stream_drop0)
{
guint i;
for (i = 0; i < G_N_ELEMENTS (rates); ++i) {
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do_perfect_stream_test (rates[i], "S8", 0.0, 0.0);
do_perfect_stream_test (rates[i], GST_AUDIO_NE (S16), 0.0, 0.0);
}
}
GST_END_TEST;
GST_START_TEST (test_perfect_stream_drop10)
{
guint i;
for (i = 0; i < G_N_ELEMENTS (rates); ++i) {
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do_perfect_stream_test (rates[i], "S8", 0.10, 0.0);
do_perfect_stream_test (rates[i], GST_AUDIO_NE (S16), 0.10, 0.0);
}
}
GST_END_TEST;
GST_START_TEST (test_perfect_stream_drop50)
{
guint i;
for (i = 0; i < G_N_ELEMENTS (rates); ++i) {
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do_perfect_stream_test (rates[i], "S8", 0.50, 0.0);
do_perfect_stream_test (rates[i], GST_AUDIO_NE (S16), 0.50, 0.0);
}
}
GST_END_TEST;
GST_START_TEST (test_perfect_stream_drop90)
{
guint i;
for (i = 0; i < G_N_ELEMENTS (rates); ++i) {
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do_perfect_stream_test (rates[i], "S8", 0.90, 0.0);
do_perfect_stream_test (rates[i], GST_AUDIO_NE (S16), 0.90, 0.0);
}
}
GST_END_TEST;
GST_START_TEST (test_perfect_stream_inject10)
{
guint i;
for (i = 0; i < G_N_ELEMENTS (rates); ++i) {
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do_perfect_stream_test (rates[i], "S8", 0.0, 0.10);
do_perfect_stream_test (rates[i], GST_AUDIO_NE (S16), 0.0, 0.10);
}
}
GST_END_TEST;
GST_START_TEST (test_perfect_stream_inject90)
{
guint i;
for (i = 0; i < G_N_ELEMENTS (rates); ++i) {
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do_perfect_stream_test (rates[i], "S8", 0.0, 0.90);
do_perfect_stream_test (rates[i], GST_AUDIO_NE (S16), 0.0, 0.90);
}
}
GST_END_TEST;
GST_START_TEST (test_perfect_stream_drop45_inject25)
{
guint i;
for (i = 0; i < G_N_ELEMENTS (rates); ++i) {
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do_perfect_stream_test (rates[i], "S8", 0.45, 0.25);
do_perfect_stream_test (rates[i], GST_AUDIO_NE (S16), 0.45, 0.25);
}
}
GST_END_TEST;
/* TODO: also do all tests with channels=1 and channels=2 */
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw,format=" GST_AUDIO_NE (F32)
",channels=1,rate=44100")
);
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw,format=" GST_AUDIO_NE (F32)
",channels=1,rate=44100")
);
GST_START_TEST (test_large_discont)
{
GstElement *audiorate;
GstCaps *caps;
GstPad *srcpad, *sinkpad;
GstBuffer *buf;
audiorate = gst_check_setup_element ("audiorate");
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caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (F32),
"layout", G_TYPE_STRING, "interleaved",
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"channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 44100, NULL);
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srcpad = gst_check_setup_src_pad (audiorate, &srctemplate);
sinkpad = gst_check_setup_sink_pad (audiorate, &sinktemplate);
gst_pad_set_active (srcpad, TRUE);
gst_check_setup_events (srcpad, audiorate, caps, GST_FORMAT_TIME);
gst_pad_set_active (sinkpad, TRUE);
fail_unless (gst_element_set_state (audiorate,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"failed to set audiorate playing");
buf = gst_buffer_new_and_alloc (4);
GST_BUFFER_TIMESTAMP (buf) = 0;
gst_pad_push (srcpad, buf);
fail_unless_equals_int (g_list_length (buffers), 1);
buf = gst_buffer_new_and_alloc (4);
GST_BUFFER_TIMESTAMP (buf) = 2 * GST_SECOND;
gst_pad_push (srcpad, buf);
/* Now we should have 3 more buffers: the one we injected, plus _two_ filler
* buffers, because the gap is > 1 second (but less than 2 seconds) */
fail_unless_equals_int (g_list_length (buffers), 4);
gst_element_set_state (audiorate, GST_STATE_NULL);
gst_caps_unref (caps);
gst_check_drop_buffers ();
gst_check_teardown_sink_pad (audiorate);
gst_check_teardown_src_pad (audiorate);
gst_object_unref (audiorate);
}
GST_END_TEST;
#define FIRST_CAPS \
"audio/x-raw,format=S16LE,layout=interleaved,rate=48000,channels=1"
#define SECOND_CAPS \
"audio/x-raw,format=S16LE,layout=interleaved,rate=8000,channels=1"
#define BUFFERS_BEFORE_CHANGE 10
#define TOTAL_BUFFERS (BUFFERS_BEFORE_CHANGE * 2)
static GList *
generate_buffers (gint from_rate, gint to_rate)
{
GQueue q = G_QUEUE_INIT;
GstBuffer *buf;
guint i;
GstClockTime pts = 0;
for (i = 0; i < BUFFERS_BEFORE_CHANGE; i++) {
buf = gst_buffer_new_allocate (NULL, 2 * from_rate / 100, NULL);
gst_buffer_memset (buf, 0, 1, gst_buffer_get_size (buf));
GST_BUFFER_PTS (buf) = pts;
GST_BUFFER_DURATION (buf) = GST_SECOND / 100;
pts += GST_BUFFER_DURATION (buf);
g_queue_push_tail (&q, buf);
}
for (; i < TOTAL_BUFFERS; i++) {
buf = gst_buffer_new_allocate (NULL, 2 * to_rate / 100, NULL);
gst_buffer_memset (buf, 0, 1, gst_buffer_get_size (buf));
GST_BUFFER_PTS (buf) = pts;
GST_BUFFER_DURATION (buf) = GST_SECOND / 100;
pts += GST_BUFFER_DURATION (buf);
g_queue_push_tail (&q, buf);
}
return q.head;
}
GST_START_TEST (test_rate_change_down)
{
GList *l, *rbufs = NULL, *bufs = NULL;
GstElement *pipeline;
GstElement *sink;
GstElement *src;
GstElement *audiorate;
GstCaps *caps1, *caps2;
int i = 0;
gint64 drop, in, out;
GstBus *bus;
caps1 = gst_caps_from_string (FIRST_CAPS);
caps2 = gst_caps_from_string (SECOND_CAPS);
bufs = generate_buffers (48000, 8000);
pipeline =
gst_parse_launch
("appsrc name=src is-live=true format=time !"
" audiorate name=audiorate ! fakesink name=sink signal-handoffs=true",
NULL);
fail_if (pipeline == NULL);
sink = gst_bin_get_by_name (GST_BIN (pipeline), "sink");
g_signal_connect (sink, "handoff", G_CALLBACK (got_buf), &rbufs);
gst_object_unref (sink);
src = gst_bin_get_by_name (GST_BIN (pipeline), "src");
gst_app_src_set_caps (GST_APP_SRC (src), caps1);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
for (l = bufs; l != NULL; l = l->next) {
if (i++ == BUFFERS_BEFORE_CHANGE) {
gst_app_src_set_caps (GST_APP_SRC (src), caps2);
}
GST_LOG ("Position: %" GST_TIME_FORMAT " Duration: %" GST_TIME_FORMAT "\n",
GST_TIME_ARGS (GST_BUFFER_PTS (l->data)),
GST_TIME_ARGS (GST_BUFFER_DURATION (l->data)));
fail_unless_equals_int (gst_app_src_push_buffer (GST_APP_SRC (src),
GST_BUFFER (l->data)), GST_FLOW_OK);
}
g_list_free (bufs);
gst_app_src_end_of_stream (GST_APP_SRC (src));
gst_object_unref (src);
/* Give some time to the appsrc loop to push the buffers */
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_message_unref (gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
GST_MESSAGE_EOS));
gst_object_unref (bus);
audiorate = gst_bin_get_by_name (GST_BIN (pipeline), "audiorate");
g_object_get (audiorate, "drop", &drop, "out", &out, "in", &in, NULL);
gst_object_unref (audiorate);
fail_unless_equals_int64 (drop, 0);
g_list_foreach (rbufs, (GFunc) gst_mini_object_unref, NULL);
g_list_free (rbufs);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
gst_caps_unref (caps1);
gst_caps_unref (caps2);
}
GST_END_TEST;
static Suite *
audiorate_suite (void)
{
Suite *s = suite_create ("audiorate");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_perfect_stream_drop0);
tcase_add_test (tc_chain, test_perfect_stream_drop10);
tcase_add_test (tc_chain, test_perfect_stream_drop50);
tcase_add_test (tc_chain, test_perfect_stream_drop90);
tcase_add_test (tc_chain, test_perfect_stream_inject10);
tcase_add_test (tc_chain, test_perfect_stream_inject90);
tcase_add_test (tc_chain, test_perfect_stream_drop45_inject25);
tcase_add_test (tc_chain, test_large_discont);
tcase_add_test (tc_chain, test_rate_change_down);
return s;
}
GST_CHECK_MAIN (audiorate);