/* GStreamer
*
* unit test for mpg123audiodec
*
* Copyright (c) 2012 Carlos Rafael Giani <dv@pseudoterminal.org>
*
* 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.
*/
#include <unistd.h>
#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>
#include <gst/app/gstappsink.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 MP2_STREAM_FILENAME "stream.mp2"
#define MP3_CBR_STREAM_FILENAME "cbr_stream.mp3"
#define MP3_VBR_STREAM_FILENAME "vbr_stream.mp3"
/* mpeg 1 layer 2 stream created with:
* gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
* "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
* avenc_mp2 bitrate=32000 ! tee name=t \
* t. ! queue ! fakesink silent=false \
* t. ! queue ! filesink location=test.mp2
*
* mpeg 1 layer 3 CBR stream created with:
* gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
* "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
* lamemp3enc encoding-engine-quality=high cbr=true target=bitrate bitrate=32 ! \
* "audio/mpeg, rate=(int)44100, channels=(int)1" ! tee name=t \
* t. ! queue ! fakesink silent=false \
* t. ! queue ! filesink location=test.mp3
*
* mpeg 1 layer 3 VBR stream created with:
* gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
* "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
* lamemp3enc encoding-engine-quality=high cbr=false target=quality quality=7 ! \
* "audio/mpeg, rate=(int)44100, channels=(int)1" ! tee name=t \
* t. ! queue ! fakesink silent=false \
* t. ! queue ! filesink location=test.mp3
*/
/* FFT test helpers taken from gst-plugins-base tests/check/audioresample.c */
#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 > -35.0) { \
GST_LOG("Found magnitude at %f : %f (peak at %f)\n", pos, mag, spot); \
return FALSE; \
} \
} \
} \
return TRUE; \
} \
static void check_main_frequency_spot_##ffttag (GstBuffer *buffer, gdouble \
expected_spot) \
{ \
GstMapInfo map; \
int num_samples; \
gdouble actual_spot; \
GstFFT##ffttag *ctx; \
GstFFT##ffttag##Complex *fftdata; \
\
gst_buffer_map (buffer, &map, GST_MAP_READ); \
\
num_samples = map.size / sizeof(type) & ~1; \
ctx = gst_fft_##ffttag2##_new (num_samples, FALSE); \
fftdata = g_new (GstFFT##ffttag##Complex, num_samples / 2 + 1); \
\
gst_fft_##ffttag2##_window (ctx, (type*)map.data, \
GST_FFT_WINDOW_HAMMING); \
gst_fft_##ffttag2##_fft (ctx, (type*)map.data, fftdata); \
\
actual_spot = find_main_frequency_spot_##ffttag (fftdata, \
num_samples / 2 + 1); \
GST_LOG ("Expected spot: %.3f actual: %.3f %f", expected_spot, actual_spot, \
fabs (expected_spot - actual_spot)); \
fail_unless (fabs (expected_spot - actual_spot) < 0.05, \
"Actual main frequency spot is too far away from expected one"); \
fail_unless (is_zero_except_##ffttag (fftdata, num_samples / 2 + 1, \
actual_spot), "One secondary peak in spectrum exceeds threshold"); \
\
gst_buffer_unmap (buffer, &map); \
\
gst_fft_##ffttag2##_free (ctx); \
g_free (fftdata); \
}
FFT_HELPERS (gint32, S32, s32, 2147483647.0);
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, format = (string) " GST_AUDIO_NE (S32))
);
static GstStaticPadTemplate layer2_srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS_ANY);
static GstStaticPadTemplate layer3_srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS_ANY);
static void
setup_input_pipeline (gchar const *stream_filename, GstElement ** pipeline,
GstElement ** appsink)
{
GstElement *source, *parser;
*pipeline = gst_pipeline_new (NULL);
source = gst_element_factory_make ("filesrc", NULL);
parser = gst_element_factory_make ("mpegaudioparse", NULL);
*appsink = gst_element_factory_make ("appsink", NULL);
gst_bin_add_many (GST_BIN (*pipeline), source, parser, *appsink, NULL);
gst_element_link_many (source, parser, *appsink, NULL);
{
char *full_filename =
g_build_filename (GST_TEST_FILES_PATH, stream_filename, NULL);
g_object_set (G_OBJECT (source), "location", full_filename, NULL);
g_free (full_filename);
}
gst_element_set_state (*pipeline, GST_STATE_PLAYING);
}
static void
cleanup_input_pipeline (GstElement * pipeline)
{
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
}
static GstElement *
setup_mpeg1layer2dec (void)
{
GstElement *mpg123audiodec;
GstCaps *caps;
GST_DEBUG ("setup_mpeg1layer2dec");
mpg123audiodec = gst_check_setup_element ("mpg123audiodec");
mysrcpad = gst_check_setup_src_pad (mpg123audiodec, &layer2_srctemplate);
mysinkpad = gst_check_setup_sink_pad (mpg123audiodec, &sinktemplate);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
/* This is necessary to trigger a set_format call in the decoder;
* fixed caps don't trigger it */
caps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, 1,
"layer", G_TYPE_INT, 2,
"rate", G_TYPE_INT, 44100,
"channels", G_TYPE_INT, 1, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
gst_check_setup_events (mysrcpad, mpg123audiodec, caps, GST_FORMAT_TIME);
gst_caps_unref (caps);
return mpg123audiodec;
}
static GstElement *
setup_mpeg1layer3dec (void)
{
GstElement *mpg123audiodec;
GstCaps *caps;
GST_DEBUG ("setup_mpeg1layer3dec");
mpg123audiodec = gst_check_setup_element ("mpg123audiodec");
mysrcpad = gst_check_setup_src_pad (mpg123audiodec, &layer3_srctemplate);
mysinkpad = gst_check_setup_sink_pad (mpg123audiodec, &sinktemplate);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
/* This is necessary to trigger a set_format call in the decoder;
* fixed caps don't trigger it */
caps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, 1,
"layer", G_TYPE_INT, 3,
"rate", G_TYPE_INT, 44100,
"channels", G_TYPE_INT, 1, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
gst_check_setup_events (mysrcpad, mpg123audiodec, caps, GST_FORMAT_TIME);
gst_caps_unref (caps);
return mpg123audiodec;
}
static void
cleanup_mpg123audiodec (GstElement * mpg123audiodec)
{
GST_DEBUG ("cleanup_mpeg1layer2dec");
gst_element_set_state (mpg123audiodec, GST_STATE_NULL);
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_src_pad (mpg123audiodec);
gst_check_teardown_sink_pad (mpg123audiodec);
gst_check_teardown_element (mpg123audiodec);
}
static void
run_decoding_test (GstElement * mpg123audiodec, gchar const *filename)
{
GstBus *bus;
unsigned int num_input_buffers, num_decoded_buffers;
gint expected_size;
GstCaps *out_caps, *caps;
GstAudioInfo audioinfo;
GstElement *input_pipeline, *input_appsink;
int i;
GstBuffer *outbuffer;
/* 440 Hz = frequency of sine wave in audio data
* 44100 Hz = sample rate
* (44100 / 2) Hz = Nyquist frequency */
static double const expected_frequency_spot = 440.0 / (44100.0 / 2.0);
fail_unless (gst_element_set_state (mpg123audiodec,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
bus = gst_bus_new ();
gst_element_set_bus (mpg123audiodec, bus);
setup_input_pipeline (filename, &input_pipeline, &input_appsink);
num_input_buffers = 0;
while (TRUE) {
GstSample *sample;
GstBuffer *input_buffer;
sample = gst_app_sink_pull_sample (GST_APP_SINK (input_appsink));
if (sample == NULL)
break;
fail_unless (GST_IS_SAMPLE (sample));
input_buffer = gst_sample_get_buffer (sample);
fail_if (input_buffer == NULL);
/* This is done to be on the safe side - docs say lifetime of the input buffer
* depends *solely* on the sample */
input_buffer = gst_buffer_copy (input_buffer);
fail_unless_equals_int (gst_pad_push (mysrcpad, input_buffer), GST_FLOW_OK);
++num_input_buffers;
gst_sample_unref (sample);
}
num_decoded_buffers = g_list_length (buffers);
/* check number of decoded buffers */
fail_unless_equals_int (num_decoded_buffers, num_input_buffers - 2);
caps = gst_pad_get_current_caps (mysinkpad);
GST_LOG ("output caps %" GST_PTR_FORMAT, caps);
fail_unless (gst_audio_info_from_caps (&audioinfo, caps),
"Getting audio info from caps failed");
/* check caps */
out_caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (S32),
"layout", G_TYPE_STRING, "interleaved",
"rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 1, NULL);
fail_unless (gst_caps_is_equal_fixed (caps, out_caps), "Incorrect out caps");
gst_caps_unref (out_caps);
gst_caps_unref (caps);
/* here, test if decoded data is a sine tone, and if the sine frequency is at the
* right spot in the spectrum */
for (i = 0; i < num_decoded_buffers; ++i) {
outbuffer = GST_BUFFER (buffers->data);
fail_if (outbuffer == NULL, "Invalid buffer retrieved");
/* MPEG 1 layer 2 uses 1152 samples per frame */
expected_size = 1152 * GST_AUDIO_INFO_BPF (&audioinfo);
fail_unless_equals_int (gst_buffer_get_size (outbuffer), expected_size);
check_main_frequency_spot_S32 (outbuffer, expected_frequency_spot);
buffers = g_list_remove (buffers, outbuffer);
gst_buffer_unref (outbuffer);
outbuffer = NULL;
}
g_list_free (buffers);
buffers = NULL;
cleanup_input_pipeline (input_pipeline);
gst_bus_set_flushing (bus, TRUE);
gst_element_set_bus (mpg123audiodec, NULL);
gst_object_unref (GST_OBJECT (bus));
}
GST_START_TEST (test_decode_mpeg1layer2)
{
GstElement *mpg123audiodec;
mpg123audiodec = setup_mpeg1layer2dec ();
run_decoding_test (mpg123audiodec, MP2_STREAM_FILENAME);
cleanup_mpg123audiodec (mpg123audiodec);
mpg123audiodec = NULL;
}
GST_END_TEST;
GST_START_TEST (test_decode_mpeg1layer3_cbr)
{
GstElement *mpg123audiodec;
mpg123audiodec = setup_mpeg1layer3dec ();
run_decoding_test (mpg123audiodec, MP3_CBR_STREAM_FILENAME);
cleanup_mpg123audiodec (mpg123audiodec);
}
GST_END_TEST;
GST_START_TEST (test_decode_mpeg1layer3_vbr)
{
GstElement *mpg123audiodec;
mpg123audiodec = setup_mpeg1layer3dec ();
run_decoding_test (mpg123audiodec, MP3_VBR_STREAM_FILENAME);
cleanup_mpg123audiodec (mpg123audiodec);
}
GST_END_TEST;
GST_START_TEST (test_decode_garbage_mpeg1layer2)
{
GstElement *mpg123audiodec;
GstBuffer *inbuffer;
GstBus *bus;
int i, num_buffers;
guint32 *tmpbuf;
mpg123audiodec = setup_mpeg1layer2dec ();
fail_unless (gst_element_set_state (mpg123audiodec,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
bus = gst_bus_new ();
/* initialize the buffer with something that is no mpeg2 */
tmpbuf = g_new (guint32, 4096);
for (i = 0; i < 4096; i++) {
tmpbuf[i] = i;
}
inbuffer = gst_buffer_new_wrapped (tmpbuf, 4096 * sizeof (guint32));
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_element_set_bus (mpg123audiodec, bus);
/* should be possible to push without problems but nothing gets decoded */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
num_buffers = g_list_length (buffers);
/* should be 0 buffers as decoding should've been impossible */
fail_unless_equals_int (num_buffers, 0);
g_list_free (buffers);
buffers = NULL;
gst_bus_set_flushing (bus, TRUE);
gst_element_set_bus (mpg123audiodec, NULL);
gst_object_unref (GST_OBJECT (bus));
cleanup_mpg123audiodec (mpg123audiodec);
mpg123audiodec = NULL;
}
GST_END_TEST;
GST_START_TEST (test_decode_garbage_mpeg1layer3)
{
GstElement *mpg123audiodec;
GstBuffer *inbuffer;
GstBus *bus;
int i, num_buffers;
guint32 *tmpbuf;
mpg123audiodec = setup_mpeg1layer3dec ();
fail_unless (gst_element_set_state (mpg123audiodec,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
bus = gst_bus_new ();
/* initialize the buffer with something that is no mpeg2 */
tmpbuf = g_new (guint32, 4096);
for (i = 0; i < 4096; i++) {
tmpbuf[i] = i;
}
inbuffer = gst_buffer_new_wrapped (tmpbuf, 4096 * sizeof (guint32));
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_element_set_bus (mpg123audiodec, bus);
/* should be possible to push without problems but nothing gets decoded */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
num_buffers = g_list_length (buffers);
/* should be 0 buffers as decoding should've been impossible */
fail_unless_equals_int (num_buffers, 0);
g_list_free (buffers);
buffers = NULL;
gst_bus_set_flushing (bus, TRUE);
gst_element_set_bus (mpg123audiodec, NULL);
gst_object_unref (GST_OBJECT (bus));
cleanup_mpg123audiodec (mpg123audiodec);
mpg123audiodec = NULL;
}
GST_END_TEST;
static gboolean
is_test_file_available (gchar const *filename)
{
gboolean ret;
gchar *full_filename;
gchar *cwd;
cwd = g_get_current_dir ();
full_filename = g_build_filename (cwd, GST_TEST_FILES_PATH, filename, NULL);
ret =
g_file_test (full_filename, G_FILE_TEST_IS_REGULAR | G_FILE_TEST_EXISTS);
g_free (full_filename);
g_free (cwd);
return ret;
}
static Suite *
mpg123audiodec_suite (void)
{
GstRegistry *registry;
Suite *s = suite_create ("mpg123audiodec");
TCase *tc_chain = tcase_create ("general");
registry = gst_registry_get ();
suite_add_tcase (s, tc_chain);
if (gst_registry_check_feature_version (registry, "filesrc",
GST_VERSION_MAJOR, GST_VERSION_MINOR, 0) &&
gst_registry_check_feature_version (registry, "mpegaudioparse",
GST_VERSION_MAJOR, GST_VERSION_MINOR, 0) &&
gst_registry_check_feature_version (registry, "appsrc",
GST_VERSION_MAJOR, GST_VERSION_MINOR, 0)) {
if (is_test_file_available (MP2_STREAM_FILENAME))
tcase_add_test (tc_chain, test_decode_mpeg1layer2);
if (is_test_file_available (MP3_CBR_STREAM_FILENAME))
tcase_add_test (tc_chain, test_decode_mpeg1layer3_cbr);
if (is_test_file_available (MP3_VBR_STREAM_FILENAME))
tcase_add_test (tc_chain, test_decode_mpeg1layer3_vbr);
}
tcase_add_test (tc_chain, test_decode_garbage_mpeg1layer2);
tcase_add_test (tc_chain, test_decode_garbage_mpeg1layer3);
return s;
}
GST_CHECK_MAIN (mpg123audiodec)