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gstreamer/tests/check/elements/audioresample.c

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/* GStreamer
*
* unit test for audioresample, based on the audioresample unit test
*
* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
* Copyright (C) <2006> Tim-Philipp Müller <tim at 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <unistd.h>
#include <gst/check/gstcheck.h>
#include <gst/audio/audio.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 RESAMPLE_CAPS_FLOAT \
"audio/x-raw-float, " \
"channels = (int) [ 1, MAX ], " \
"rate = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) { 32, 64 }"
#define RESAMPLE_CAPS_INT \
"audio/x-raw-int, " \
"channels = (int) [ 1, MAX ], " \
"rate = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 16, " \
"depth = (int) 16, " \
"signed = (bool) TRUE"
#define RESAMPLE_CAPS_TEMPLATE_STRING \
RESAMPLE_CAPS_FLOAT " ; " \
RESAMPLE_CAPS_INT
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (RESAMPLE_CAPS_TEMPLATE_STRING)
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (RESAMPLE_CAPS_TEMPLATE_STRING)
);
static GstElement *
setup_audioresample (int channels, int inrate, int outrate, int width,
gboolean fp)
{
GstElement *audioresample;
GstCaps *caps;
GstStructure *structure;
GST_DEBUG ("setup_audioresample");
audioresample = gst_check_setup_element ("audioresample");
if (fp)
caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
else
caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
structure = gst_caps_get_structure (caps, 0);
gst_structure_set (structure, "channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, inrate, "width", G_TYPE_INT, width, NULL);
if (!fp)
gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
fail_unless (gst_caps_is_fixed (caps));
fail_unless (gst_element_set_state (audioresample,
GST_STATE_PAUSED) == GST_STATE_CHANGE_SUCCESS,
"could not set to paused");
mysrcpad = gst_check_setup_src_pad (audioresample, &srctemplate, caps);
gst_pad_set_caps (mysrcpad, caps);
gst_caps_unref (caps);
if (fp)
caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
else
caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
structure = gst_caps_get_structure (caps, 0);
gst_structure_set (structure, "channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, outrate, "width", G_TYPE_INT, width, NULL);
if (!fp)
gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
fail_unless (gst_caps_is_fixed (caps));
mysinkpad = gst_check_setup_sink_pad (audioresample, &sinktemplate, caps);
/* this installs a getcaps func that will always return the caps we set
* later */
gst_pad_set_caps (mysinkpad, caps);
gst_pad_use_fixed_caps (mysinkpad);
gst_pad_set_active (mysinkpad, TRUE);
gst_pad_set_active (mysrcpad, TRUE);
gst_caps_unref (caps);
return audioresample;
}
static void
cleanup_audioresample (GstElement * audioresample)
{
GST_DEBUG ("cleanup_audioresample");
fail_unless (gst_element_set_state (audioresample,
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_src_pad (audioresample);
gst_check_teardown_sink_pad (audioresample);
gst_check_teardown_element (audioresample);
gst_check_drop_buffers ();
}
static void
fail_unless_perfect_stream (void)
{
guint64 timestamp = 0L, duration = 0L;
guint64 offset = 0L, offset_end = 0L;
GList *l;
GstBuffer *buffer;
for (l = buffers; l; l = l->next) {
buffer = GST_BUFFER (l->data);
ASSERT_BUFFER_REFCOUNT (buffer, "buffer", 1);
GST_DEBUG ("buffer timestamp %" G_GUINT64_FORMAT ", duration %"
G_GUINT64_FORMAT " offset %" G_GUINT64_FORMAT " offset_end %"
G_GUINT64_FORMAT,
GST_BUFFER_TIMESTAMP (buffer),
GST_BUFFER_DURATION (buffer),
GST_BUFFER_OFFSET (buffer), GST_BUFFER_OFFSET_END (buffer));
fail_unless_equals_uint64 (timestamp, GST_BUFFER_TIMESTAMP (buffer));
fail_unless_equals_uint64 (offset, GST_BUFFER_OFFSET (buffer));
duration = GST_BUFFER_DURATION (buffer);
offset_end = GST_BUFFER_OFFSET_END (buffer);
timestamp += duration;
offset = offset_end;
gst_buffer_unref (buffer);
}
g_list_free (buffers);
buffers = NULL;
}
/* this tests that the output is a perfect stream if the input is */
static void
test_perfect_stream_instance (int inrate, int outrate, int samples,
int numbuffers)
{
GstElement *audioresample;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
guint64 offset = 0;
int i, j;
gint16 *p;
audioresample = setup_audioresample (2, inrate, outrate, 16, FALSE);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
fail_unless (gst_element_set_state (audioresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
for (j = 1; j <= numbuffers; ++j) {
inbuffer = gst_buffer_new_and_alloc (samples * 4);
GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (samples, inrate);
GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1);
GST_BUFFER_OFFSET (inbuffer) = offset;
offset += samples;
GST_BUFFER_OFFSET_END (inbuffer) = offset;
gst_buffer_set_caps (inbuffer, caps);
p = (gint16 *) GST_BUFFER_DATA (inbuffer);
/* create a 16 bit signed ramp */
for (i = 0; i < samples; ++i) {
*p = -32767 + i * (65535 / samples);
++p;
*p = -32767 + i * (65535 / samples);
++p;
}
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... but it ends up being collected on the global buffer list */
fail_unless_equals_int (g_list_length (buffers), j);
}
/* FIXME: we should make audioresample handle eos by flushing out the last
* samples, which will give us one more, small, buffer */
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
fail_unless_perfect_stream ();
/* cleanup */
gst_caps_unref (caps);
cleanup_audioresample (audioresample);
}
/* make sure that outgoing buffers are contiguous in timestamp/duration and
* offset/offsetend
*/
GST_START_TEST (test_perfect_stream)
{
/* integral scalings */
test_perfect_stream_instance (48000, 24000, 500, 20);
test_perfect_stream_instance (48000, 12000, 500, 20);
test_perfect_stream_instance (12000, 24000, 500, 20);
test_perfect_stream_instance (12000, 48000, 500, 20);
/* non-integral scalings */
test_perfect_stream_instance (44100, 8000, 500, 20);
test_perfect_stream_instance (8000, 44100, 500, 20);
/* wacky scalings */
test_perfect_stream_instance (12345, 54321, 500, 20);
test_perfect_stream_instance (101, 99, 500, 20);
}
GST_END_TEST;
/* this tests that the output is a correct discontinuous stream
* if the input is; ie input drops in time come out the same way */
static void
test_discont_stream_instance (int inrate, int outrate, int samples,
int numbuffers)
{
GstElement *audioresample;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
GstClockTime ints;
int i, j;
gint16 *p;
GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d",
inrate, outrate, samples, numbuffers);
audioresample = setup_audioresample (2, inrate, outrate, 16, FALSE);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
fail_unless (gst_element_set_state (audioresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
for (j = 1; j <= numbuffers; ++j) {
inbuffer = gst_buffer_new_and_alloc (samples * 4);
GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate;
/* "drop" half the buffers */
ints = GST_BUFFER_DURATION (inbuffer) * 2 * (j - 1);
GST_BUFFER_TIMESTAMP (inbuffer) = ints;
GST_BUFFER_OFFSET (inbuffer) = (j - 1) * 2 * samples;
GST_BUFFER_OFFSET_END (inbuffer) = j * 2 * samples + samples;
gst_buffer_set_caps (inbuffer, caps);
p = (gint16 *) GST_BUFFER_DATA (inbuffer);
/* create a 16 bit signed ramp */
for (i = 0; i < samples; ++i) {
*p = -32767 + i * (65535 / samples);
++p;
*p = -32767 + i * (65535 / samples);
++p;
}
GST_DEBUG ("Sending Buffer time:%" G_GUINT64_FORMAT " duration:%"
G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (inbuffer),
GST_BUFFER_DURATION (inbuffer), GST_BUFFER_IS_DISCONT (inbuffer),
GST_BUFFER_OFFSET (inbuffer), GST_BUFFER_OFFSET_END (inbuffer));
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* check if the timestamp of the pushed buffer matches the incoming one */
outbuffer = g_list_nth_data (buffers, g_list_length (buffers) - 1);
fail_if (outbuffer == NULL);
fail_unless_equals_uint64 (ints, GST_BUFFER_TIMESTAMP (outbuffer));
GST_DEBUG ("Got Buffer time:%" G_GUINT64_FORMAT " duration:%"
G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (outbuffer),
GST_BUFFER_DURATION (outbuffer), GST_BUFFER_IS_DISCONT (outbuffer),
GST_BUFFER_OFFSET (outbuffer), GST_BUFFER_OFFSET_END (outbuffer));
if (j > 1) {
fail_unless (GST_BUFFER_IS_DISCONT (outbuffer),
"expected discont for buffer #%d", j);
}
}
/* cleanup */
gst_caps_unref (caps);
cleanup_audioresample (audioresample);
}
GST_START_TEST (test_discont_stream)
{
/* integral scalings */
test_discont_stream_instance (48000, 24000, 5000, 20);
test_discont_stream_instance (48000, 12000, 5000, 20);
test_discont_stream_instance (12000, 24000, 5000, 20);
test_discont_stream_instance (12000, 48000, 5000, 20);
/* non-integral scalings */
test_discont_stream_instance (44100, 8000, 5000, 20);
test_discont_stream_instance (8000, 44100, 5000, 20);
/* wacky scalings */
test_discont_stream_instance (12345, 54321, 5000, 20);
test_discont_stream_instance (101, 99, 5000, 20);
}
GST_END_TEST;
GST_START_TEST (test_reuse)
{
GstElement *audioresample;
GstEvent *newseg;
GstBuffer *inbuffer;
GstCaps *caps;
audioresample = setup_audioresample (1, 9343, 48000, 16, FALSE);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
fail_unless (gst_element_set_state (audioresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
inbuffer = gst_buffer_new_and_alloc (9343 * 4);
memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
GST_BUFFER_OFFSET (inbuffer) = 0;
gst_buffer_set_caps (inbuffer, caps);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... but it ends up being collected on the global buffer list */
fail_unless_equals_int (g_list_length (buffers), 1);
/* now reset and try again ... */
fail_unless (gst_element_set_state (audioresample,
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
fail_unless (gst_element_set_state (audioresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
inbuffer = gst_buffer_new_and_alloc (9343 * 4);
memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
GST_BUFFER_OFFSET (inbuffer) = 0;
gst_buffer_set_caps (inbuffer, caps);
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... it also ends up being collected on the global buffer list. If we
* now have more than 2 buffers, then audioresample probably didn't clean
* up its internal buffer properly and tried to push the remaining samples
* when it got the second NEWSEGMENT event */
fail_unless_equals_int (g_list_length (buffers), 2);
cleanup_audioresample (audioresample);
gst_caps_unref (caps);
}
GST_END_TEST;
GST_START_TEST (test_shutdown)
{
GstElement *pipeline, *src, *cf1, *ar, *cf2, *sink;
GstCaps *caps;
guint i;
/* create pipeline, force audioresample to actually resample */
pipeline = gst_pipeline_new (NULL);
src = gst_check_setup_element ("audiotestsrc");
cf1 = gst_check_setup_element ("capsfilter");
ar = gst_check_setup_element ("audioresample");
cf2 = gst_check_setup_element ("capsfilter");
g_object_set (cf2, "name", "capsfilter2", NULL);
sink = gst_check_setup_element ("fakesink");
caps =
gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT, 11025, NULL);
g_object_set (cf1, "caps", caps, NULL);
gst_caps_unref (caps);
caps =
gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT, 48000, NULL);
g_object_set (cf2, "caps", caps, NULL);
gst_caps_unref (caps);
/* don't want to sync against the clock, the more throughput the better */
g_object_set (src, "is-live", FALSE, NULL);
g_object_set (sink, "sync", FALSE, NULL);
gst_bin_add_many (GST_BIN (pipeline), src, cf1, ar, cf2, sink, NULL);
fail_if (!gst_element_link_many (src, cf1, ar, cf2, sink, NULL));
/* now, wait until pipeline is running and then shut it down again; repeat */
for (i = 0; i < 20; ++i) {
gst_element_set_state (pipeline, GST_STATE_PAUSED);
gst_element_get_state (pipeline, NULL, NULL, -1);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
g_usleep (100);
gst_element_set_state (pipeline, GST_STATE_NULL);
}
gst_object_unref (pipeline);
}
GST_END_TEST;
static GstFlowReturn
live_switch_alloc_only_48000 (GstPad * pad, guint64 offset,
guint size, GstCaps * caps, GstBuffer ** buf)
{
GstStructure *structure;
gint rate;
gint channels;
GstCaps *desired;
structure = gst_caps_get_structure (caps, 0);
fail_unless (gst_structure_get_int (structure, "rate", &rate));
fail_unless (gst_structure_get_int (structure, "channels", &channels));
if (rate < 48000)
return GST_FLOW_NOT_NEGOTIATED;
desired = gst_caps_copy (caps);
gst_caps_set_simple (desired, "rate", G_TYPE_INT, 48000, NULL);
*buf = gst_buffer_new_and_alloc (channels * 48000);
gst_buffer_set_caps (*buf, desired);
gst_caps_unref (desired);
return GST_FLOW_OK;
}
static GstCaps *
live_switch_get_sink_caps (GstPad * pad)
{
GstCaps *result;
result = gst_caps_copy (GST_PAD_CAPS (pad));
gst_caps_set_simple (result,
"rate", GST_TYPE_INT_RANGE, 48000, G_MAXINT, NULL);
return result;
}
static void
live_switch_push (int rate, GstCaps * caps)
{
GstBuffer *inbuffer;
GstCaps *desired;
GList *l;
desired = gst_caps_copy (caps);
gst_caps_set_simple (desired, "rate", G_TYPE_INT, rate, NULL);
gst_pad_set_caps (mysrcpad, desired);
fail_unless (gst_pad_alloc_buffer_and_set_caps (mysrcpad,
GST_BUFFER_OFFSET_NONE, rate * 4, desired, &inbuffer) == GST_FLOW_OK);
/* When the basetransform hits the non-configured case it always
* returns a buffer with exactly the same caps as we requested so the actual
* renegotiation (if needed) will be done in the _chain*/
fail_unless (inbuffer != NULL);
GST_DEBUG ("desired: %" GST_PTR_FORMAT ".... got: %" GST_PTR_FORMAT,
desired, GST_BUFFER_CAPS (inbuffer));
fail_unless (gst_caps_is_equal (desired, GST_BUFFER_CAPS (inbuffer)));
memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
GST_BUFFER_OFFSET (inbuffer) = 0;
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... but it ends up being collected on the global buffer list */
fail_unless_equals_int (g_list_length (buffers), 1);
for (l = buffers; l; l = l->next) {
GstBuffer *buffer = GST_BUFFER (l->data);
gst_buffer_unref (buffer);
}
g_list_free (buffers);
buffers = NULL;
gst_caps_unref (desired);
}
GST_START_TEST (test_live_switch)
{
GstElement *audioresample;
GstEvent *newseg;
GstCaps *caps;
audioresample = setup_audioresample (4, 48000, 48000, 16, FALSE);
/* Let the sinkpad act like something that can only handle things of
* rate 48000- and can only allocate buffers for that rate, but if someone
* tries to get a buffer with a rate higher then 48000 tries to renegotiate
* */
gst_pad_set_bufferalloc_function (mysinkpad, live_switch_alloc_only_48000);
gst_pad_set_getcaps_function (mysinkpad, live_switch_get_sink_caps);
gst_pad_use_fixed_caps (mysrcpad);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
fail_unless (gst_element_set_state (audioresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
/* downstream can provide the requested rate, a buffer alloc will be passed
* on */
live_switch_push (48000, caps);
/* Downstream can never accept this rate, buffer alloc isn't passed on */
live_switch_push (40000, caps);
/* Downstream can provide the requested rate but will re-negotiate */
live_switch_push (50000, caps);
cleanup_audioresample (audioresample);
gst_caps_unref (caps);
}
GST_END_TEST;
#ifndef GST_DISABLE_PARSE
static GMainLoop *loop;
static gint messages = 0;
static void
element_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
{
gchar *s;
s = gst_structure_to_string (gst_message_get_structure (message));
GST_DEBUG ("Received message: %s", s);
g_free (s);
messages++;
}
static void
eos_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
{
GST_DEBUG ("Received eos");
g_main_loop_quit (loop);
}
static void
test_pipeline (gint width, gboolean fp, gint inrate, gint outrate, gint quality)
{
GstElement *pipeline;
GstBus *bus;
GError *error = NULL;
gchar *pipe_str;
pipe_str =
g_strdup_printf
("audiotestsrc num-buffers=10 ! audioconvert ! audio/x-raw-%s,rate=%d,width=%d,channels=2 ! audioresample quality=%d ! audio/x-raw-%s,rate=%d,width=%d ! identity check-imperfect-timestamp=TRUE ! fakesink",
(fp) ? "float" : "int", inrate, width, quality, (fp) ? "float" : "int",
outrate, width);
pipeline = gst_parse_launch (pipe_str, &error);
fail_unless (pipeline != NULL, "Error parsing pipeline: %s",
error ? error->message : "(invalid error)");
g_free (pipe_str);
bus = gst_element_get_bus (pipeline);
fail_if (bus == NULL);
gst_bus_add_signal_watch (bus);
g_signal_connect (bus, "message::element", (GCallback) element_message_cb,
NULL);
g_signal_connect (bus, "message::eos", (GCallback) eos_message_cb, NULL);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
/* run until we receive EOS */
loop = g_main_loop_new (NULL, FALSE);
g_main_loop_run (loop);
g_main_loop_unref (loop);
loop = NULL;
gst_element_set_state (pipeline, GST_STATE_NULL);
fail_if (messages > 0, "Received imperfect timestamp messages");
gst_object_unref (pipeline);
}
GST_START_TEST (test_pipelines)
{
gint quality;
/* Test qualities 0, 5 and 10 */
for (quality = 0; quality < 11; quality += 5) {
GST_DEBUG ("Checking with quality %d", quality);
test_pipeline (8, FALSE, 44100, 48000, quality);
test_pipeline (8, FALSE, 48000, 44100, quality);
test_pipeline (16, FALSE, 44100, 48000, quality);
test_pipeline (16, FALSE, 48000, 44100, quality);
test_pipeline (24, FALSE, 44100, 48000, quality);
test_pipeline (24, FALSE, 48000, 44100, quality);
test_pipeline (32, FALSE, 44100, 48000, quality);
test_pipeline (32, FALSE, 48000, 44100, quality);
test_pipeline (32, TRUE, 44100, 48000, quality);
test_pipeline (32, TRUE, 48000, 44100, quality);
test_pipeline (64, TRUE, 44100, 48000, quality);
test_pipeline (64, TRUE, 48000, 44100, quality);
}
}
GST_END_TEST;
GST_START_TEST (test_preference_passthrough)
{
GstStateChangeReturn ret;
GstElement *pipeline, *src;
GstStructure *s;
GstMessage *msg;
GstCaps *caps;
GstPad *pad;
GstBus *bus;
GError *error = NULL;
gint rate = 0;
pipeline = gst_parse_launch ("audiotestsrc num-buffers=1 name=src ! "
"audioresample ! audio/x-raw-int,channels=1,width=16,depth=16,"
"endianness=BYTE_ORDER,signed=true,rate=8000 ! "
"fakesink can-activate-pull=false", &error);
fail_unless (pipeline != NULL, "Error parsing pipeline: %s",
error ? error->message : "(invalid error)");
ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
fail_unless_equals_int (ret, GST_STATE_CHANGE_ASYNC);
/* run until we receive EOS */
bus = gst_element_get_bus (pipeline);
fail_if (bus == NULL);
msg = gst_bus_timed_pop_filtered (bus, -1, GST_MESSAGE_EOS);
gst_message_unref (msg);
gst_object_unref (bus);
src = gst_bin_get_by_name (GST_BIN (pipeline), "src");
fail_unless (src != NULL);
pad = gst_element_get_static_pad (src, "src");
fail_unless (pad != NULL);
caps = gst_pad_get_negotiated_caps (pad);
GST_LOG ("negotiated audiotestsrc caps: %" GST_PTR_FORMAT, caps);
fail_unless (caps != NULL);
s = gst_caps_get_structure (caps, 0);
fail_unless (gst_structure_get_int (s, "rate", &rate));
/* there's no need to resample, audiotestsrc supports any rate, so make
* sure audioresample provided upstream with the right caps to negotiate
* this correctly */
fail_unless_equals_int (rate, 8000);
gst_caps_unref (caps);
gst_object_unref (pad);
gst_object_unref (src);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
}
GST_END_TEST;
#endif
static void
_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
{
GMainLoop *loop = user_data;
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR:
case GST_MESSAGE_WARNING:
g_assert_not_reached ();
break;
case GST_MESSAGE_EOS:
g_main_loop_quit (loop);
break;
default:
break;
}
}
typedef struct
{
guint64 latency;
GstClockTime in_ts;
GstClockTime next_out_ts;
guint64 next_out_off;
guint64 in_buffer_count, out_buffer_count;
} TimestampDriftCtx;
static void
fakesink_handoff_cb (GstElement * object, GstBuffer * buffer, GstPad * pad,
gpointer user_data)
{
TimestampDriftCtx *ctx = user_data;
ctx->out_buffer_count++;
if (ctx->latency == GST_CLOCK_TIME_NONE) {
ctx->latency = 1000 - GST_BUFFER_SIZE (buffer) / 8;
}
/* Check if we have a perfectly timestampped stream */
if (ctx->next_out_ts != GST_CLOCK_TIME_NONE)
fail_unless (ctx->next_out_ts == GST_BUFFER_TIMESTAMP (buffer),
"expected timestamp %" GST_TIME_FORMAT " got timestamp %"
GST_TIME_FORMAT, GST_TIME_ARGS (ctx->next_out_ts),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
/* Check if we have a perfectly offsetted stream */
fail_unless (GST_BUFFER_OFFSET_END (buffer) ==
GST_BUFFER_OFFSET (buffer) + GST_BUFFER_SIZE (buffer) / 8,
"expected offset end %" G_GUINT64_FORMAT " got offset end %"
G_GUINT64_FORMAT,
GST_BUFFER_OFFSET (buffer) + GST_BUFFER_SIZE (buffer) / 8,
GST_BUFFER_OFFSET_END (buffer));
if (ctx->next_out_off != GST_BUFFER_OFFSET_NONE) {
fail_unless (GST_BUFFER_OFFSET (buffer) == ctx->next_out_off,
"expected offset %" G_GUINT64_FORMAT " got offset %" G_GUINT64_FORMAT,
ctx->next_out_off, GST_BUFFER_OFFSET (buffer));
}
if (ctx->in_buffer_count != ctx->out_buffer_count) {
GST_INFO ("timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
}
if (ctx->in_ts != GST_CLOCK_TIME_NONE && ctx->in_buffer_count > 1
&& ctx->in_buffer_count == ctx->out_buffer_count) {
fail_unless (GST_BUFFER_TIMESTAMP (buffer) ==
ctx->in_ts - gst_util_uint64_scale_round (ctx->latency, GST_SECOND,
4096),
"expected output timestamp %" GST_TIME_FORMAT " (%" G_GUINT64_FORMAT
") got output timestamp %" GST_TIME_FORMAT " (%" G_GUINT64_FORMAT ")",
GST_TIME_ARGS (ctx->in_ts - gst_util_uint64_scale_round (ctx->latency,
GST_SECOND, 4096)),
ctx->in_ts - gst_util_uint64_scale_round (ctx->latency, GST_SECOND,
4096), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_BUFFER_TIMESTAMP (buffer));
}
ctx->next_out_ts =
GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer);
ctx->next_out_off = GST_BUFFER_OFFSET_END (buffer);
}
static void
identity_handoff_cb (GstElement * object, GstBuffer * buffer,
gpointer user_data)
{
TimestampDriftCtx *ctx = user_data;
ctx->in_ts = GST_BUFFER_TIMESTAMP (buffer);
ctx->in_buffer_count++;
}
GST_START_TEST (test_timestamp_drift)
{
TimestampDriftCtx ctx =
{ GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE,
GST_BUFFER_OFFSET_NONE, 0, 0
};
GstElement *pipeline;
GstElement *audiotestsrc, *capsfilter1, *identity, *audioresample,
*capsfilter2, *fakesink;
GstBus *bus;
GMainLoop *loop;
GstCaps *caps;
pipeline = gst_pipeline_new ("pipeline");
fail_unless (pipeline != NULL);
audiotestsrc = gst_element_factory_make ("audiotestsrc", "src");
fail_unless (audiotestsrc != NULL);
g_object_set (G_OBJECT (audiotestsrc), "num-buffers", 10000,
"samplesperbuffer", 4000, NULL);
capsfilter1 = gst_element_factory_make ("capsfilter", "capsfilter1");
fail_unless (capsfilter1 != NULL);
caps =
gst_caps_from_string
("audio/x-raw-float, channels=1, width=64, rate=16384");
g_object_set (G_OBJECT (capsfilter1), "caps", caps, NULL);
gst_caps_unref (caps);
identity = gst_element_factory_make ("identity", "identity");
fail_unless (identity != NULL);
g_object_set (G_OBJECT (identity), "sync", FALSE, "signal-handoffs", TRUE,
NULL);
g_signal_connect (identity, "handoff", (GCallback) identity_handoff_cb, &ctx);
audioresample = gst_element_factory_make ("audioresample", "resample");
fail_unless (audioresample != NULL);
capsfilter2 = gst_element_factory_make ("capsfilter", "capsfilter2");
fail_unless (capsfilter2 != NULL);
caps =
gst_caps_from_string
("audio/x-raw-float, channels=1, width=64, rate=4096");
g_object_set (G_OBJECT (capsfilter2), "caps", caps, NULL);
gst_caps_unref (caps);
fakesink = gst_element_factory_make ("fakesink", "sink");
fail_unless (fakesink != NULL);
g_object_set (G_OBJECT (fakesink), "sync", FALSE, "async", FALSE,
"signal-handoffs", TRUE, NULL);
g_signal_connect (fakesink, "handoff", (GCallback) fakesink_handoff_cb, &ctx);
gst_bin_add_many (GST_BIN (pipeline), audiotestsrc, capsfilter1, identity,
audioresample, capsfilter2, fakesink, NULL);
fail_unless (gst_element_link_many (audiotestsrc, capsfilter1, identity,
audioresample, capsfilter2, fakesink, NULL));
loop = g_main_loop_new (NULL, FALSE);
bus = gst_element_get_bus (pipeline);
gst_bus_add_signal_watch (bus);
g_signal_connect (bus, "message", (GCallback) _message_cb, loop);
fail_unless (gst_element_set_state (pipeline,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
g_main_loop_run (loop);
fail_unless (gst_element_set_state (pipeline,
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS);
g_main_loop_unref (loop);
gst_object_unref (pipeline);
} GST_END_TEST;
static Suite *
audioresample_suite (void)
{
Suite *s = suite_create ("audioresample");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_perfect_stream);
tcase_add_test (tc_chain, test_discont_stream);
tcase_add_test (tc_chain, test_reuse);
tcase_add_test (tc_chain, test_shutdown);
tcase_add_test (tc_chain, test_live_switch);
tcase_add_test (tc_chain, test_timestamp_drift);
#ifndef GST_DISABLE_PARSE
tcase_set_timeout (tc_chain, 360);
tcase_add_test (tc_chain, test_pipelines);
tcase_add_test (tc_chain, test_preference_passthrough);
#endif
return s;
}
GST_CHECK_MAIN (audioresample);
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