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audioresample: add FFT based checks
Send a few simple tones through audioresample and check that the main frequency spot is the same for the input and the resampled output. https://bugzilla.gnome.org/show_bug.cgi?id=656392
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22cc529409
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505bae099e
2 changed files with 194 additions and 2 deletions
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@ -400,6 +400,7 @@ elements_audioresample_CFLAGS = \
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$(AM_CFLAGS)
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$(AM_CFLAGS)
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elements_audioresample_LDADD = \
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elements_audioresample_LDADD = \
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$(top_builddir)/gst-libs/gst/fft/libgstfft-@GST_MAJORMINOR@.la \
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$(top_builddir)/gst-libs/gst/audio/libgstaudio-@GST_MAJORMINOR@.la \
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$(top_builddir)/gst-libs/gst/audio/libgstaudio-@GST_MAJORMINOR@.la \
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$(top_builddir)/gst-libs/gst/interfaces/libgstinterfaces-@GST_MAJORMINOR@.la \
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$(top_builddir)/gst-libs/gst/interfaces/libgstinterfaces-@GST_MAJORMINOR@.la \
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$(GST_BASE_LIBS) \
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$(GST_BASE_LIBS) \
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@ -27,6 +27,12 @@
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#include <gst/audio/audio.h>
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#include <gst/audio/audio.h>
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#include <gst/fft/gstfft.h>
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#include <gst/fft/gstffts16.h>
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#include <gst/fft/gstffts32.h>
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#include <gst/fft/gstfftf32.h>
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#include <gst/fft/gstfftf64.h>
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/* For ease of programming we use globals to keep refs for our floating
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/* For ease of programming we use globals to keep refs for our floating
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* src and sink pads we create; otherwise we always have to do get_pad,
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* src and sink pads we create; otherwise we always have to do get_pad,
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* get_peer, and then remove references in every test function */
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* get_peer, and then remove references in every test function */
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@ -44,8 +50,8 @@ static GstPad *mysrcpad, *mysinkpad;
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"channels = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, MAX ], " \
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"rate = (int) [ 1, MAX ], " \
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"rate = (int) [ 1, MAX ], " \
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"endianness = (int) BYTE_ORDER, " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) 16, " \
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"width = (int) { 16, 32 }, " \
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"depth = (int) 16, " \
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"depth = (int) { 16, 32 }, " \
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"signed = (bool) TRUE"
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"signed = (bool) TRUE"
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#define RESAMPLE_CAPS_TEMPLATE_STRING \
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#define RESAMPLE_CAPS_TEMPLATE_STRING \
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@ -900,6 +906,190 @@ GST_START_TEST (test_timestamp_drift)
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} GST_END_TEST;
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} GST_END_TEST;
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#define FFT_HELPERS(type,ffttag,ffttag2,scale); \
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static gdouble magnitude##ffttag (const GstFFT##ffttag##Complex *c) \
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{ \
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gdouble mag = (gdouble) c->r * (gdouble) c->r; \
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mag += (gdouble) c->i * (gdouble) c->i; \
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mag /= scale * scale; \
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mag = 10.0 * log10 (mag); \
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return mag; \
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} \
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static gdouble find_main_frequency_spot_##ffttag (const GstFFT##ffttag##Complex *v, \
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int elements) \
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{ \
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int i; \
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gdouble maxmag = -9999; \
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int maxidx = 0; \
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for (i=0; i<elements; ++i) { \
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gdouble mag = magnitude##ffttag (v+i); \
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if (mag > maxmag) { \
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maxmag = mag; \
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maxidx = i; \
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} \
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} \
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return maxidx / (gdouble) elements; \
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} \
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static gboolean is_zero_except_##ffttag (const GstFFT##ffttag##Complex *v, int elements, \
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gdouble spot) \
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{ \
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int i; \
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for (i=0; i<elements; ++i) { \
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gdouble pos = i / (gdouble) elements; \
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gdouble mag = magnitude##ffttag (v+i); \
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if (fabs (pos - spot) > 0.01) { \
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if (mag > -55.0) { \
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return FALSE; \
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} \
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} \
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} \
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return TRUE; \
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} \
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static void compare_ffts_##ffttag (const GstBuffer *inbuffer, const GstBuffer *outbuffer) \
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{ \
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int insamples = GST_BUFFER_SIZE (inbuffer) / sizeof(type) & ~1; \
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int outsamples = GST_BUFFER_SIZE (outbuffer) / sizeof(type) & ~1; \
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gdouble inspot, outspot; \
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\
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GstFFT##ffttag *inctx = gst_fft_##ffttag2##_new (insamples, FALSE); \
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GstFFT##ffttag##Complex *in = g_new (GstFFT##ffttag##Complex, insamples / 2 + 1); \
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GstFFT##ffttag *outctx = gst_fft_##ffttag2##_new (outsamples, FALSE); \
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GstFFT##ffttag##Complex *out = g_new (GstFFT##ffttag##Complex, outsamples / 2 + 1); \
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\
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gst_fft_##ffttag2##_window (inctx, (type*)GST_BUFFER_DATA (inbuffer), \
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GST_FFT_WINDOW_HAMMING); \
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gst_fft_##ffttag2##_fft (inctx, (type*)GST_BUFFER_DATA (inbuffer), in); \
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gst_fft_##ffttag2##_window (outctx, (type*)GST_BUFFER_DATA (outbuffer), \
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GST_FFT_WINDOW_HAMMING); \
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gst_fft_##ffttag2##_fft (outctx, (type*)GST_BUFFER_DATA (outbuffer), out); \
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\
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inspot = find_main_frequency_spot_##ffttag (in, insamples / 2 + 1); \
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outspot = find_main_frequency_spot_##ffttag (out, outsamples / 2 + 1); \
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GST_LOG ("Spots are %.3f and %.3f", inspot, outspot); \
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fail_unless (fabs (outspot - inspot) < 0.05); \
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fail_unless (is_zero_except_##ffttag (in, insamples / 2 + 1, inspot)); \
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fail_unless (is_zero_except_##ffttag (out, outsamples / 2 + 1, outspot)); \
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\
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gst_fft_##ffttag2##_free (inctx); \
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gst_fft_##ffttag2##_free (outctx); \
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g_free (in); \
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g_free (out); \
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}
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FFT_HELPERS (float, F32, f32, 2048.0f);
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FFT_HELPERS (double, F64, f64, 2048.0);
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FFT_HELPERS (gint16, S16, s16, 32767.0);
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FFT_HELPERS (gint32, S32, s32, 2147483647.0);
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#define FILL_BUFFER(type, desc, value); \
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static void init_##type##_##desc (GstBuffer *buffer) \
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{ \
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type *ptr = (type *) GST_BUFFER_DATA (buffer); \
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int i, nsamples = GST_BUFFER_SIZE (buffer) / sizeof (type); \
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for (i = 0; i < nsamples; ++i) { \
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*ptr++ = value; \
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} \
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}
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FILL_BUFFER (float, silence, 0.0f);
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FILL_BUFFER (double, silence, 0.0);
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FILL_BUFFER (gint16, silence, 0);
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FILL_BUFFER (gint32, silence, 0);
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FILL_BUFFER (float, sine, sinf (i * 0.01f));
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FILL_BUFFER (float, sine2, sinf (i * 1.8f));
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FILL_BUFFER (double, sine, sin (i * 0.01));
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FILL_BUFFER (double, sine2, sin (i * 1.8));
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FILL_BUFFER (gint16, sine, (gint16) (32767 * sinf (i * 0.01f)));
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FILL_BUFFER (gint16, sine2, (gint16) (32767 * sinf (i * 1.8f)));
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FILL_BUFFER (gint32, sine, (gint32) (2147483647 * sinf (i * 0.01f)));
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FILL_BUFFER (gint32, sine2, (gint32) (2147483647 * sinf (i * 1.8f)));
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static void
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run_fft_pipeline (int inrate, int outrate, int quality, int width, gboolean fp,
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void (*init) (GstBuffer *),
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void (*compare_ffts) (const GstBuffer *, const GstBuffer *))
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{
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GstElement *audioresample;
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GstBuffer *inbuffer, *outbuffer;
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GstCaps *caps;
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const int nsamples = 2048;
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audioresample = setup_audioresample (1, inrate, outrate, width, fp);
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fail_unless (audioresample != NULL);
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g_object_set (audioresample, "quality", quality, NULL);
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caps = gst_pad_get_negotiated_caps (mysrcpad);
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fail_unless (gst_caps_is_fixed (caps));
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fail_unless (gst_element_set_state (audioresample,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
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"could not set to playing");
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inbuffer = gst_buffer_new_and_alloc (nsamples * width / 8);
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GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (nsamples, inrate);
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GST_BUFFER_TIMESTAMP (inbuffer) = 0;
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gst_buffer_set_caps (inbuffer, caps);
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gst_buffer_ref (inbuffer);
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(*init) (inbuffer);
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/* pushing gives away my reference ... */
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fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
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/* ... but it ends up being collected on the global buffer list */
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fail_unless_equals_int (g_list_length (buffers), 1);
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/* retrieve out buffer */
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fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
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fail_unless (gst_element_set_state (audioresample,
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GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to null");
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(*compare_ffts) (inbuffer, outbuffer);
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/* cleanup */
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gst_buffer_unref (inbuffer);
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cleanup_audioresample (audioresample);
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}
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GST_START_TEST (test_fft)
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{
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int quality;
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size_t f0, f1;
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static const int frequencies[] =
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{ 8000, 16000, 44100, 48000, 128000, 12345, 54321 };
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/* audioresample uses a mixed float/double code path for floats with quality>8, make sure we test it */
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for (quality = 0; quality <= 10; quality += 5) {
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for (f0 = 0; f0 < G_N_ELEMENTS (frequencies); ++f0) {
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for (f1 = 0; f1 < G_N_ELEMENTS (frequencies); ++f1) {
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run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, TRUE,
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&init_float_silence, &compare_ffts_F32);
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run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, TRUE,
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&init_float_sine, &compare_ffts_F32);
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run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, TRUE,
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&init_float_sine2, &compare_ffts_F32);
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run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64, TRUE,
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&init_double_silence, &compare_ffts_F64);
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run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64, TRUE,
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&init_double_sine, &compare_ffts_F64);
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run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64, TRUE,
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&init_double_sine2, &compare_ffts_F64);
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run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16, FALSE,
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&init_gint16_silence, &compare_ffts_S16);
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run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16, FALSE,
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&init_gint16_sine, &compare_ffts_S16);
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run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16, FALSE,
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&init_gint16_sine2, &compare_ffts_S16);
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run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, FALSE,
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&init_gint32_silence, &compare_ffts_S32);
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run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, FALSE,
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&init_gint32_sine, &compare_ffts_S32);
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run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, FALSE,
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&init_gint32_sine2, &compare_ffts_S32);
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}
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}
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}
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}
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GST_END_TEST;
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static Suite *
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static Suite *
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audioresample_suite (void)
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audioresample_suite (void)
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{
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{
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@ -913,6 +1103,7 @@ audioresample_suite (void)
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tcase_add_test (tc_chain, test_shutdown);
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tcase_add_test (tc_chain, test_shutdown);
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tcase_add_test (tc_chain, test_live_switch);
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tcase_add_test (tc_chain, test_live_switch);
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tcase_add_test (tc_chain, test_timestamp_drift);
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tcase_add_test (tc_chain, test_timestamp_drift);
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tcase_add_test (tc_chain, test_fft);
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#ifndef GST_DISABLE_PARSE
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#ifndef GST_DISABLE_PARSE
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tcase_set_timeout (tc_chain, 360);
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tcase_set_timeout (tc_chain, 360);
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