/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/check/gstcheck.h>
#include <gst/audio/audio.h>
#include <gst/fft/gstfft.h>
#include <gst/fft/gstffts16.h>
#include <gst/fft/gstffts32.h>
#include <gst/fft/gstfftf32.h>
#include <gst/fft/gstfftf64.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;
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define FORMATS "{ F32LE, F64LE, S16LE, S32LE }"
#else
#define FORMATS "{ F32BE, F64BE, S16BE, S32BE }"
#endif
#define RESAMPLE_CAPS \
"audio/x-raw, " \
"format = (string) "FORMATS", " \
"channels = (int) [ 1, MAX ], " \
"rate = (int) [ 1, MAX ], " \
"layout = (string) interleaved"
static GstElement *
setup_audioresample (int channels, guint64 mask, int inrate, int outrate,
const gchar * format)
{
GstPadTemplate *sinktemplate;
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (RESAMPLE_CAPS)
);
GstElement *audioresample;
GstCaps *caps;
GstStructure *structure;
GST_DEBUG ("setup_audioresample");
audioresample = gst_check_setup_element ("audioresample");
caps = gst_caps_from_string (RESAMPLE_CAPS);
structure = gst_caps_get_structure (caps, 0);
gst_structure_set (structure, "channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, inrate, "format", G_TYPE_STRING, format,
"channel-mask", GST_TYPE_BITMASK, mask, 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);
gst_pad_set_active (mysrcpad, TRUE);
gst_check_setup_events (mysrcpad, audioresample, caps, GST_FORMAT_TIME);
gst_caps_unref (caps);
caps = gst_caps_from_string (RESAMPLE_CAPS);
structure = gst_caps_get_structure (caps, 0);
gst_structure_set (structure, "channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, outrate, "format", G_TYPE_STRING, format, NULL);
fail_unless (gst_caps_is_fixed (caps));
sinktemplate =
gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, caps);
mysinkpad =
gst_check_setup_sink_pad_from_template (audioresample, sinktemplate);
gst_pad_set_active (mysinkpad, TRUE);
/* this installs a getcaps func that will always return the caps we set
* later */
gst_pad_use_fixed_caps (mysinkpad);
gst_caps_unref (caps);
gst_object_unref (sinktemplate);
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;
GstMapInfo map;
gint16 *p;
audioresample =
setup_audioresample (2, 0x3, inrate, outrate, GST_AUDIO_NE (S16));
caps = gst_pad_get_current_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_map (inbuffer, &map, GST_MAP_WRITE);
p = (gint16 *) map.data;
/* 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_buffer_unmap (inbuffer, &map);
/* 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;
GstMapInfo map;
gint16 *p;
GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d",
inrate, outrate, samples, numbuffers);
audioresample =
setup_audioresample (2, 3, inrate, outrate, GST_AUDIO_NE (S16));
caps = gst_pad_get_current_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_map (inbuffer, &map, GST_MAP_WRITE);
p = (gint16 *) map.data;
/* 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_buffer_unmap (inbuffer, &map);
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;
GstSegment segment;
audioresample = setup_audioresample (1, 0, 9343, 48000, GST_AUDIO_NE (S16));
caps = gst_pad_get_current_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");
gst_segment_init (&segment, GST_FORMAT_TIME);
newseg = gst_event_new_segment (&segment);
fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
inbuffer = gst_buffer_new_and_alloc (9343 * 4);
gst_buffer_memset (inbuffer, 0, 0, 9343 * 4);
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);
/* 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_segment (&segment);
fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
inbuffer = gst_buffer_new_and_alloc (9343 * 4);
gst_buffer_memset (inbuffer, 0, 0, 9343 * 4);
GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
GST_BUFFER_OFFSET (inbuffer) = 0;
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", "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", "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;
#if 0
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_make_writable (gst_pad_get_current_caps (pad));
gst_caps_set_simple (result,
"rate", GST_TYPE_INT_RANGE, 48000, G_MAXINT, NULL);
return result;
}
#endif
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);
#if 0
fail_unless (gst_pad_alloc_buffer_and_set_caps (mysrcpad,
GST_BUFFER_OFFSET_NONE, rate * 4, desired, &inbuffer) == GST_FLOW_OK);
#endif
inbuffer = gst_buffer_new_and_alloc (rate * 4);
gst_buffer_memset (inbuffer, 0, 0, rate * 4);
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;
GstSegment segment;
audioresample =
setup_audioresample (4, 0xf, 48000, 48000, GST_AUDIO_NE (S16));
/* 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_current_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");
gst_segment_init (&segment, GST_FORMAT_TIME);
newseg = gst_event_new_segment (&segment);
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 (const gchar * format, 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,format=%s,rate=%d,channels=2 ! audioresample quality=%d ! audio/x-raw,format=%s,rate=%d ! identity check-imperfect-timestamp=TRUE ! fakesink",
format, inrate, quality, format, outrate);
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);
gst_bus_remove_signal_watch (bus);
gst_object_unref (bus);
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 ("S8", 44100, 48000, quality);
test_pipeline ("S8", 48000, 44100, quality);
test_pipeline (GST_AUDIO_NE (S16), 44100, 48000, quality);
test_pipeline (GST_AUDIO_NE (S16), 48000, 44100, quality);
test_pipeline (GST_AUDIO_NE (S24), 44100, 48000, quality);
test_pipeline (GST_AUDIO_NE (S24), 48000, 44100, quality);
test_pipeline (GST_AUDIO_NE (S32), 44100, 48000, quality);
test_pipeline (GST_AUDIO_NE (S32), 48000, 44100, quality);
test_pipeline (GST_AUDIO_NE (F32), 44100, 48000, quality);
test_pipeline (GST_AUDIO_NE (F32), 48000, 44100, quality);
test_pipeline (GST_AUDIO_NE (F64), 44100, 48000, quality);
test_pipeline (GST_AUDIO_NE (F64), 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,format=" GST_AUDIO_NE (S16) ",channels=1,"
"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_current_caps (pad);
GST_LOG ("current 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_get_size (buffer) / 8;
}
/* Check if we have a perfectly timestamped 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_get_size (buffer) / 8,
"expected offset end %" G_GUINT64_FORMAT " got offset end %"
G_GUINT64_FORMAT,
GST_BUFFER_OFFSET (buffer) + gst_buffer_get_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, format=" GST_AUDIO_NE (F64)
", channels=1, 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, format=" GST_AUDIO_NE (F64)
", channels=1, 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_bus_remove_signal_watch (bus);
gst_object_unref (bus);
gst_object_unref (pipeline);
} GST_END_TEST;
#define FFT_HELPERS(type,ffttag,ffttag2,scale); \
static gdouble magnitude##ffttag (const GstFFT##ffttag##Complex *c) \
{ \
gdouble mag = (gdouble) c->r * (gdouble) c->r; \
mag += (gdouble) c->i * (gdouble) c->i; \
mag /= scale * scale; \
mag = 10.0 * log10 (mag); \
return mag; \
} \
static gdouble find_main_frequency_spot_##ffttag (const GstFFT##ffttag##Complex *v, \
int elements) \
{ \
int i; \
gdouble maxmag = -9999; \
int maxidx = 0; \
for (i=0; i<elements; ++i) { \
gdouble mag = magnitude##ffttag (v+i); \
if (mag > maxmag) { \
maxmag = mag; \
maxidx = i; \
} \
} \
return maxidx / (gdouble) elements; \
} \
static gboolean is_zero_except_##ffttag (const GstFFT##ffttag##Complex *v, int elements, \
gdouble spot) \
{ \
int i; \
for (i=0; i<elements; ++i) { \
gdouble pos = i / (gdouble) elements; \
gdouble mag = magnitude##ffttag (v+i); \
if (fabs (pos - spot) > 0.01) { \
if (mag > -55.0) { \
return FALSE; \
} \
} \
} \
return TRUE; \
} \
static void compare_ffts_##ffttag (GstBuffer *inbuffer, GstBuffer *outbuffer) \
{ \
GstMapInfo inmap, outmap; \
int insamples, outsamples; \
gdouble inspot, outspot; \
GstFFT##ffttag *inctx, *outctx; \
GstFFT##ffttag##Complex *in, *out; \
\
gst_buffer_map (inbuffer, &inmap, GST_MAP_READ); \
gst_buffer_map (outbuffer, &outmap, GST_MAP_READWRITE); \
\
insamples = inmap.size / sizeof(type) & ~1; \
outsamples = outmap.size / sizeof(type) & ~1; \
inctx = gst_fft_##ffttag2##_new (insamples, FALSE); \
outctx = gst_fft_##ffttag2##_new (outsamples, FALSE); \
in = g_new (GstFFT##ffttag##Complex, insamples / 2 + 1); \
out = g_new (GstFFT##ffttag##Complex, outsamples / 2 + 1); \
\
gst_fft_##ffttag2##_window (inctx, (type*)inmap.data, \
GST_FFT_WINDOW_HAMMING); \
gst_fft_##ffttag2##_fft (inctx, (type*)inmap.data, in); \
gst_fft_##ffttag2##_window (outctx, (type*)outmap.data, \
GST_FFT_WINDOW_HAMMING); \
gst_fft_##ffttag2##_fft (outctx, (type*)outmap.data, out); \
\
inspot = find_main_frequency_spot_##ffttag (in, insamples / 2 + 1); \
outspot = find_main_frequency_spot_##ffttag (out, outsamples / 2 + 1); \
GST_LOG ("Spots are %.3f and %.3f", inspot, outspot); \
fail_unless (fabs (outspot - inspot) < 0.05); \
fail_unless (is_zero_except_##ffttag (in, insamples / 2 + 1, inspot)); \
fail_unless (is_zero_except_##ffttag (out, outsamples / 2 + 1, outspot)); \
\
gst_buffer_unmap (inbuffer, &inmap); \
gst_buffer_unmap (outbuffer, &outmap); \
\
gst_fft_##ffttag2##_free (inctx); \
gst_fft_##ffttag2##_free (outctx); \
g_free (in); \
g_free (out); \
}
FFT_HELPERS (float, F32, f32, 2048.0f);
FFT_HELPERS (double, F64, f64, 2048.0);
FFT_HELPERS (gint16, S16, s16, 32767.0);
FFT_HELPERS (gint32, S32, s32, 2147483647.0);
#define FILL_BUFFER(type, desc, value); \
static void init_##type##_##desc (GstBuffer *buffer) \
{ \
GstMapInfo map; \
type *ptr; \
int i, nsamples; \
gst_buffer_map (buffer, &map, GST_MAP_WRITE); \
ptr = (type *)map.data; \
nsamples = map.size / sizeof (type); \
for (i = 0; i < nsamples; ++i) { \
*ptr++ = value; \
} \
gst_buffer_unmap (buffer, &map); \
}
FILL_BUFFER (float, silence, 0.0f);
FILL_BUFFER (double, silence, 0.0);
FILL_BUFFER (gint16, silence, 0);
FILL_BUFFER (gint32, silence, 0);
FILL_BUFFER (float, sine, sinf (i * 0.01f));
FILL_BUFFER (float, sine2, sinf (i * 1.8f));
FILL_BUFFER (double, sine, sin (i * 0.01));
FILL_BUFFER (double, sine2, sin (i * 1.8));
FILL_BUFFER (gint16, sine, (gint16) (32767 * sinf (i * 0.01f)));
FILL_BUFFER (gint16, sine2, (gint16) (32767 * sinf (i * 1.8f)));
FILL_BUFFER (gint32, sine, (gint32) (2147483647 * sinf (i * 0.01f)));
FILL_BUFFER (gint32, sine2, (gint32) (2147483647 * sinf (i * 1.8f)));
static void
run_fft_pipeline (int inrate, int outrate, int quality, int width,
const gchar * format, void (*init) (GstBuffer *),
void (*compare_ffts) (GstBuffer *, GstBuffer *))
{
GstElement *audioresample;
GstBuffer *inbuffer, *outbuffer;
const int nsamples = 2048;
audioresample = setup_audioresample (1, 0, inrate, outrate, format);
fail_unless (audioresample != NULL);
g_object_set (audioresample, "quality", quality, NULL);
fail_unless (gst_element_set_state (audioresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
inbuffer = gst_buffer_new_and_alloc (nsamples * width / 8);
GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (nsamples, inrate);
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
(*init) (inbuffer);
gst_buffer_ref (inbuffer);
/* 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);
/* retrieve out buffer */
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
fail_unless (gst_element_set_state (audioresample,
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to null");
if (inbuffer == outbuffer)
gst_buffer_unref (inbuffer);
(*compare_ffts) (inbuffer, outbuffer);
/* cleanup */
cleanup_audioresample (audioresample);
}
GST_START_TEST (test_fft)
{
int quality;
size_t f0, f1;
static const int frequencies[] =
{ 8000, 16000, 44100, 48000, 128000, 12345, 54321 };
/* audioresample uses a mixed float/double code path for floats with quality>8, make sure we test it */
for (quality = 0; quality <= 10; quality += 5) {
for (f0 = 0; f0 < G_N_ELEMENTS (frequencies); ++f0) {
for (f1 = 0; f1 < G_N_ELEMENTS (frequencies); ++f1) {
run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
GST_AUDIO_NE (F32), &init_float_silence, &compare_ffts_F32);
run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
GST_AUDIO_NE (F32), &init_float_sine, &compare_ffts_F32);
run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
GST_AUDIO_NE (F32), &init_float_sine2, &compare_ffts_F32);
run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64,
GST_AUDIO_NE (F64), &init_double_silence, &compare_ffts_F64);
run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64,
GST_AUDIO_NE (F64), &init_double_sine, &compare_ffts_F64);
run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64,
GST_AUDIO_NE (F64), &init_double_sine2, &compare_ffts_F64);
run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16,
GST_AUDIO_NE (S16), &init_gint16_silence, &compare_ffts_S16);
run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16,
GST_AUDIO_NE (S16), &init_gint16_sine, &compare_ffts_S16);
run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16,
GST_AUDIO_NE (S16), &init_gint16_sine2, &compare_ffts_S16);
run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
GST_AUDIO_NE (S32), &init_gint32_silence, &compare_ffts_S32);
run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
GST_AUDIO_NE (S32), &init_gint32_sine, &compare_ffts_S32);
run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
GST_AUDIO_NE (S32), &init_gint32_sine2, &compare_ffts_S32);
}
}
}
}
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);
tcase_add_test (tc_chain, test_fft);
#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);