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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-09 13:48:55 +00:00
Co-authored-by: Sebastian Dröge <sebastian@centricular.com> Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1028>
651 lines
23 KiB
C
651 lines
23 KiB
C
/* GStreamer
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*
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* unit test for mpg123audiodec
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*
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* Copyright (c) 2012 Carlos Rafael Giani <dv@pseudoterminal.org>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <unistd.h>
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#include <gst/check/gstcheck.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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#include <gst/app/gstappsink.h>
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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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* get_peer, and then remove references in every test function */
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static GstPad *mysrcpad, *mysinkpad;
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#define MP2_STREAM_FILENAME "stream.mp2"
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#define MP3_CBR_STREAM_FILENAME "cbr_stream.mp3"
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#define MP3_VBR_STREAM_FILENAME "vbr_stream.mp3"
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#define MP3_GAPLESS_STREAM_FILENAME "sine-1009ms-1ch-32000hz-gapless-with-lame-tag.mp3"
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/* mpeg 1 layer 2 stream created with:
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* gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
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* "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
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* avenc_mp2 bitrate=32000 ! tee name=t \
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* t. ! queue ! fakesink silent=false \
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* t. ! queue ! filesink location=test.mp2
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*
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* mpeg 1 layer 3 CBR stream created with:
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* gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
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* "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
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* lamemp3enc encoding-engine-quality=high cbr=true target=bitrate bitrate=32 ! \
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* "audio/mpeg, rate=(int)44100, channels=(int)1" ! tee name=t \
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* t. ! queue ! fakesink silent=false \
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* t. ! queue ! filesink location=test.mp3
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*
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* mpeg 1 layer 3 VBR stream created with:
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* gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
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* "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
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* lamemp3enc encoding-engine-quality=high cbr=false target=quality quality=7 ! \
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* "audio/mpeg, rate=(int)44100, channels=(int)1" ! tee name=t \
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* t. ! queue ! fakesink silent=false \
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* t. ! queue ! filesink location=test.mp3
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*/
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/* FFT test helpers taken from gst-plugins-base tests/check/audioresample.c */
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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 ( \
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const GstFFT##ffttag##Complex *v, 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, \
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int elements, 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 > -35.0) { \
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GST_LOG("Found magnitude at %f : %f (peak at %f)\n", pos, mag, spot); \
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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 check_main_frequency_spot_##ffttag (GstBuffer *buffer, gdouble \
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expected_spot) \
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{ \
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GstMapInfo map; \
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int num_samples; \
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gdouble actual_spot; \
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GstFFT##ffttag *ctx; \
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GstFFT##ffttag##Complex *fftdata; \
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\
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gst_buffer_map (buffer, &map, GST_MAP_READ); \
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\
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num_samples = map.size / sizeof(type) & ~1; \
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ctx = gst_fft_##ffttag2##_new (num_samples, FALSE); \
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fftdata = g_new (GstFFT##ffttag##Complex, num_samples / 2 + 1); \
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\
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gst_fft_##ffttag2##_window (ctx, (type*)map.data, \
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GST_FFT_WINDOW_HAMMING); \
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gst_fft_##ffttag2##_fft (ctx, (type*)map.data, fftdata); \
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\
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actual_spot = find_main_frequency_spot_##ffttag (fftdata, \
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num_samples / 2 + 1); \
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GST_LOG ("Expected spot: %.3f actual: %.3f %f", expected_spot, actual_spot, \
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fabs (expected_spot - actual_spot)); \
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fail_unless (fabs (expected_spot - actual_spot) < 0.05, \
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"Actual main frequency spot is too far away from expected one"); \
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fail_unless (is_zero_except_##ffttag (fftdata, num_samples / 2 + 1, \
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actual_spot), "One secondary peak in spectrum exceeds threshold"); \
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\
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gst_buffer_unmap (buffer, &map); \
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\
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gst_fft_##ffttag2##_free (ctx); \
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g_free (fftdata); \
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}
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FFT_HELPERS (gint32, S32, s32, 2147483647.0);
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static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, format = (string) " GST_AUDIO_NE (S32))
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);
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static GstStaticPadTemplate layer2_srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS_ANY);
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static GstStaticPadTemplate layer3_srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS_ANY);
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static void
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setup_input_pipeline (gchar const *stream_filename, GstElement ** pipeline,
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GstElement ** appsink)
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{
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GstElement *source, *parser;
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*pipeline = gst_pipeline_new (NULL);
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source = gst_element_factory_make ("filesrc", NULL);
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parser = gst_element_factory_make ("mpegaudioparse", NULL);
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*appsink = gst_element_factory_make ("appsink", NULL);
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gst_bin_add_many (GST_BIN (*pipeline), source, parser, *appsink, NULL);
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gst_element_link_many (source, parser, *appsink, NULL);
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{
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char *full_filename =
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g_build_filename (GST_TEST_FILES_PATH, stream_filename, NULL);
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g_object_set (G_OBJECT (source), "location", full_filename, NULL);
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g_free (full_filename);
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}
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gst_element_set_state (*pipeline, GST_STATE_PLAYING);
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}
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static void
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cleanup_input_pipeline (GstElement * pipeline)
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{
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gst_element_set_state (pipeline, GST_STATE_NULL);
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gst_object_unref (pipeline);
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}
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static GstElement *
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setup_mpeg1layer2dec (void)
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{
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GstElement *mpg123audiodec;
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GstCaps *caps;
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GST_DEBUG ("setup_mpeg1layer2dec");
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mpg123audiodec = gst_check_setup_element ("mpg123audiodec");
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mysrcpad = gst_check_setup_src_pad (mpg123audiodec, &layer2_srctemplate);
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mysinkpad = gst_check_setup_sink_pad (mpg123audiodec, &sinktemplate);
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gst_pad_set_active (mysrcpad, TRUE);
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gst_pad_set_active (mysinkpad, TRUE);
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/* This is necessary to trigger a set_format call in the decoder;
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* fixed caps don't trigger it */
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caps = gst_caps_new_simple ("audio/mpeg",
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"mpegversion", G_TYPE_INT, 1,
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"layer", G_TYPE_INT, 2,
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"rate", G_TYPE_INT, 44100,
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"channels", G_TYPE_INT, 1, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
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gst_check_setup_events (mysrcpad, mpg123audiodec, caps, GST_FORMAT_TIME);
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gst_caps_unref (caps);
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return mpg123audiodec;
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}
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static GstElement *
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setup_mpeg1layer3dec (gint sample_rate)
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{
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GstElement *mpg123audiodec;
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GstCaps *caps;
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GST_DEBUG ("setup_mpeg1layer3dec");
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mpg123audiodec = gst_check_setup_element ("mpg123audiodec");
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mysrcpad = gst_check_setup_src_pad (mpg123audiodec, &layer3_srctemplate);
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mysinkpad = gst_check_setup_sink_pad (mpg123audiodec, &sinktemplate);
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gst_pad_set_active (mysrcpad, TRUE);
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gst_pad_set_active (mysinkpad, TRUE);
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/* This is necessary to trigger a set_format call in the decoder;
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* fixed caps don't trigger it */
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caps = gst_caps_new_simple ("audio/mpeg",
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"mpegversion", G_TYPE_INT, 1,
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"layer", G_TYPE_INT, 3,
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"rate", G_TYPE_INT, sample_rate,
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"channels", G_TYPE_INT, 1, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
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gst_check_setup_events (mysrcpad, mpg123audiodec, caps, GST_FORMAT_TIME);
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gst_caps_unref (caps);
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return mpg123audiodec;
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}
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static void
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cleanup_mpg123audiodec (GstElement * mpg123audiodec)
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{
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GST_DEBUG ("cleanup_mpeg1layer2dec");
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gst_element_set_state (mpg123audiodec, GST_STATE_NULL);
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gst_pad_set_active (mysrcpad, FALSE);
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gst_pad_set_active (mysinkpad, FALSE);
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gst_check_teardown_src_pad (mpg123audiodec);
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gst_check_teardown_sink_pad (mpg123audiodec);
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gst_check_teardown_element (mpg123audiodec);
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}
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static void
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run_decoding_test (GstElement * mpg123audiodec, gchar const *filename)
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{
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GstBus *bus;
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unsigned int num_input_buffers, num_decoded_buffers;
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gint expected_size;
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GstCaps *out_caps, *caps;
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GstAudioInfo audioinfo;
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GstElement *input_pipeline, *input_appsink;
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int i;
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GstBuffer *outbuffer;
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/* 440 Hz = frequency of sine wave in audio data
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* 44100 Hz = sample rate
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* (44100 / 2) Hz = Nyquist frequency */
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static double const expected_frequency_spot = 440.0 / (44100.0 / 2.0);
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fail_unless (gst_element_set_state (mpg123audiodec,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
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"could not set to playing");
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bus = gst_bus_new ();
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gst_element_set_bus (mpg123audiodec, bus);
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setup_input_pipeline (filename, &input_pipeline, &input_appsink);
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num_input_buffers = 0;
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while (TRUE) {
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GstSample *sample;
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GstBuffer *input_buffer;
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sample = gst_app_sink_pull_sample (GST_APP_SINK (input_appsink));
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if (sample == NULL)
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break;
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fail_unless (GST_IS_SAMPLE (sample));
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input_buffer = gst_sample_get_buffer (sample);
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fail_if (input_buffer == NULL);
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/* This is done to be on the safe side - docs say lifetime of the input buffer
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* depends *solely* on the sample */
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input_buffer = gst_buffer_ref (input_buffer);
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fail_unless_equals_int (gst_pad_push (mysrcpad, input_buffer), GST_FLOW_OK);
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++num_input_buffers;
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gst_sample_unref (sample);
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}
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num_decoded_buffers = g_list_length (buffers);
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/* check number of decoded buffers */
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fail_unless_equals_int (num_decoded_buffers, num_input_buffers);
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caps = gst_pad_get_current_caps (mysinkpad);
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GST_LOG ("output caps %" GST_PTR_FORMAT, caps);
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fail_unless (gst_audio_info_from_caps (&audioinfo, caps),
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"Getting audio info from caps failed");
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/* check caps */
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out_caps = gst_caps_new_simple ("audio/x-raw",
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"format", G_TYPE_STRING, GST_AUDIO_NE (S32),
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"layout", G_TYPE_STRING, "interleaved",
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"rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 1, NULL);
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fail_unless (gst_caps_is_equal_fixed (caps, out_caps), "Incorrect out caps");
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gst_caps_unref (out_caps);
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gst_caps_unref (caps);
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/* here, test if decoded data is a sine tone, and if the sine frequency is at the
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* right spot in the spectrum */
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for (i = 0; i < num_decoded_buffers; ++i) {
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fail_if (buffers == NULL);
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outbuffer = GST_BUFFER (buffers->data);
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fail_if (outbuffer == NULL, "Invalid buffer retrieved");
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/* MPEG 1 layer 2 uses 1152 samples per frame */
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expected_size = 1152 * GST_AUDIO_INFO_BPF (&audioinfo);
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fail_unless_equals_int (gst_buffer_get_size (outbuffer), expected_size);
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check_main_frequency_spot_S32 (outbuffer, expected_frequency_spot);
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buffers = g_list_delete_link (buffers, buffers);
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gst_buffer_unref (outbuffer);
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outbuffer = NULL;
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}
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fail_unless (buffers == NULL);
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cleanup_input_pipeline (input_pipeline);
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gst_bus_set_flushing (bus, TRUE);
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gst_element_set_bus (mpg123audiodec, NULL);
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gst_object_unref (GST_OBJECT (bus));
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}
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GST_START_TEST (test_decode_mpeg1layer2)
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{
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GstElement *mpg123audiodec;
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mpg123audiodec = setup_mpeg1layer2dec ();
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run_decoding_test (mpg123audiodec, MP2_STREAM_FILENAME);
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cleanup_mpg123audiodec (mpg123audiodec);
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mpg123audiodec = NULL;
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}
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GST_END_TEST;
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GST_START_TEST (test_decode_mpeg1layer3_cbr)
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{
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GstElement *mpg123audiodec;
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mpg123audiodec = setup_mpeg1layer3dec (44100);
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run_decoding_test (mpg123audiodec, MP3_CBR_STREAM_FILENAME);
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cleanup_mpg123audiodec (mpg123audiodec);
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}
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GST_END_TEST;
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GST_START_TEST (test_decode_mpeg1layer3_vbr)
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{
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GstElement *mpg123audiodec;
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mpg123audiodec = setup_mpeg1layer3dec (44100);
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run_decoding_test (mpg123audiodec, MP3_VBR_STREAM_FILENAME);
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cleanup_mpg123audiodec (mpg123audiodec);
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}
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GST_END_TEST;
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GST_START_TEST (test_decode_mpeg1layer3_gapless)
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{
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GstBus *bus;
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guint num_decoded_buffers;
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guint num_decoded_pcm_frames;
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GstCaps *out_caps, *caps;
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GstAudioInfo audioinfo;
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GstElement *input_pipeline, *input_appsink;
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int i;
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GstBuffer *outbuffer;
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GstElement *mpg123audiodec;
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/* 440 Hz = frequency of sine wave in audio data
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* 32000 Hz = sample rate
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* (32000 / 2) Hz = Nyquist frequency */
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static double const expected_frequency_spot = 440.0 / (32000.0 / 2.0);
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mpg123audiodec = setup_mpeg1layer3dec (32000);
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fail_unless (gst_element_set_state (mpg123audiodec,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
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"could not set to playing");
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bus = gst_bus_new ();
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gst_element_set_bus (mpg123audiodec, bus);
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setup_input_pipeline (MP3_GAPLESS_STREAM_FILENAME, &input_pipeline,
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&input_appsink);
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while (TRUE) {
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GstSample *sample;
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GstBuffer *input_buffer;
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sample = gst_app_sink_pull_sample (GST_APP_SINK (input_appsink));
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if (sample == NULL)
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break;
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fail_unless (GST_IS_SAMPLE (sample));
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input_buffer = gst_sample_get_buffer (sample);
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fail_if (input_buffer == NULL);
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/* This is done to be on the safe side - docs say lifetime of the input buffer
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* depends *solely* on the sample */
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input_buffer = gst_buffer_ref (input_buffer);
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|
fail_unless_equals_int (gst_pad_push (mysrcpad, input_buffer), GST_FLOW_OK);
|
|
|
|
gst_sample_unref (sample);
|
|
}
|
|
|
|
num_decoded_buffers = g_list_length (buffers);
|
|
|
|
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, 32000, "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);
|
|
|
|
/* This is the main check. We see how many PCM frames got decoded
|
|
* in total. If the amount is not what we expected, then gapless
|
|
* decoding failed, because padding samples have to be omitted
|
|
* in order for the playback to be really gapless. */
|
|
num_decoded_pcm_frames = 0;
|
|
for (i = 0; i < num_decoded_buffers; ++i) {
|
|
guint num_frames;
|
|
|
|
fail_if (buffers == NULL);
|
|
outbuffer = GST_BUFFER (buffers->data);
|
|
fail_if (outbuffer == NULL, "Invalid buffer retrieved");
|
|
|
|
num_frames =
|
|
gst_buffer_get_size (outbuffer) / GST_AUDIO_INFO_BPF (&audioinfo);
|
|
num_decoded_pcm_frames += num_frames;
|
|
|
|
/* Don't check the first frame for a sine wave, because it will
|
|
* unavoidably have a discontinuity at the beginning, causing the
|
|
* spectrum to be filled with additional peaks, so the FFT check
|
|
* will detect false positives. */
|
|
if (i != 0)
|
|
check_main_frequency_spot_S32 (outbuffer, expected_frequency_spot);
|
|
|
|
buffers = g_list_delete_link (buffers, buffers);
|
|
gst_buffer_unref (outbuffer);
|
|
outbuffer = NULL;
|
|
}
|
|
|
|
fail_unless_equals_int (num_decoded_pcm_frames, 32288);
|
|
fail_unless (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));
|
|
|
|
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 (44100);
|
|
|
|
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;
|
|
|
|
if (g_path_is_absolute (GST_TEST_FILES_PATH)) {
|
|
full_filename = g_build_filename (GST_TEST_FILES_PATH, filename, NULL);
|
|
} else {
|
|
gchar *cwd = g_get_current_dir ();
|
|
full_filename = g_build_filename (cwd, GST_TEST_FILES_PATH, filename, NULL);
|
|
g_free (cwd);
|
|
}
|
|
ret =
|
|
g_file_test (full_filename, G_FILE_TEST_IS_REGULAR | G_FILE_TEST_EXISTS);
|
|
g_free (full_filename);
|
|
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);
|
|
if (is_test_file_available (MP3_GAPLESS_STREAM_FILENAME))
|
|
tcase_add_test (tc_chain, test_decode_mpeg1layer3_gapless);
|
|
}
|
|
tcase_add_test (tc_chain, test_decode_garbage_mpeg1layer2);
|
|
tcase_add_test (tc_chain, test_decode_garbage_mpeg1layer3);
|
|
|
|
return s;
|
|
}
|
|
|
|
|
|
GST_CHECK_MAIN (mpg123audiodec)
|