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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1113 lines
41 KiB
C
1113 lines
41 KiB
C
/* GStreamer
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*
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* unit test for audioresample, based on the audioresample unit test
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*
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* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
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* Copyright (C) <2006> Tim-Philipp Müller <tim at centricular net>
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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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/* 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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#if G_BYTE_ORDER == G_LITTLE_ENDIAN
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#define FORMATS "{ F32LE, F64LE, S16LE, S32LE }"
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#else
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#define FORMATS "{ F32BE, F64BE, S16BE, S32BE }"
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#endif
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#define RESAMPLE_CAPS \
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"audio/x-raw, " \
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"format = (string) "FORMATS", " \
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"channels = (int) [ 1, MAX ], " \
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"rate = (int) [ 1, MAX ], " \
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"layout = (string) interleaved"
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static GstElement *
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setup_audioresample (int channels, guint64 mask, int inrate, int outrate,
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const gchar * format)
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{
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GstPadTemplate *sinktemplate;
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static GstStaticPadTemplate 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 (RESAMPLE_CAPS)
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);
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GstElement *audioresample;
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GstCaps *caps;
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GstStructure *structure;
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GST_DEBUG ("setup_audioresample");
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audioresample = gst_check_setup_element ("audioresample");
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caps = gst_caps_from_string (RESAMPLE_CAPS);
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_set (structure, "channels", G_TYPE_INT, channels,
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"rate", G_TYPE_INT, inrate, "format", G_TYPE_STRING, format,
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"channel-mask", GST_TYPE_BITMASK, mask, NULL);
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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_PAUSED) == GST_STATE_CHANGE_SUCCESS,
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"could not set to paused");
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mysrcpad = gst_check_setup_src_pad (audioresample, &srctemplate);
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gst_pad_set_active (mysrcpad, TRUE);
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gst_check_setup_events (mysrcpad, audioresample, caps, GST_FORMAT_TIME);
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gst_caps_unref (caps);
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caps = gst_caps_from_string (RESAMPLE_CAPS);
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_set (structure, "channels", G_TYPE_INT, channels,
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"rate", G_TYPE_INT, outrate, "format", G_TYPE_STRING, format, NULL);
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fail_unless (gst_caps_is_fixed (caps));
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sinktemplate =
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gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, caps);
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mysinkpad =
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gst_check_setup_sink_pad_from_template (audioresample, sinktemplate);
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gst_pad_set_active (mysinkpad, TRUE);
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/* this installs a getcaps func that will always return the caps we set
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* later */
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gst_pad_use_fixed_caps (mysinkpad);
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gst_caps_unref (caps);
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gst_object_unref (sinktemplate);
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return audioresample;
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}
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static void
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cleanup_audioresample (GstElement * audioresample)
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{
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GST_DEBUG ("cleanup_audioresample");
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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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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 (audioresample);
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gst_check_teardown_sink_pad (audioresample);
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gst_check_teardown_element (audioresample);
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gst_check_drop_buffers ();
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}
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static void
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fail_unless_perfect_stream (void)
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{
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guint64 timestamp = 0L, duration = 0L;
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guint64 offset = 0L, offset_end = 0L;
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GList *l;
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GstBuffer *buffer;
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for (l = buffers; l; l = l->next) {
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buffer = GST_BUFFER (l->data);
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ASSERT_BUFFER_REFCOUNT (buffer, "buffer", 1);
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GST_DEBUG ("buffer timestamp %" G_GUINT64_FORMAT ", duration %"
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G_GUINT64_FORMAT " offset %" G_GUINT64_FORMAT " offset_end %"
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G_GUINT64_FORMAT,
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GST_BUFFER_TIMESTAMP (buffer),
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GST_BUFFER_DURATION (buffer),
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GST_BUFFER_OFFSET (buffer), GST_BUFFER_OFFSET_END (buffer));
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fail_unless_equals_uint64 (timestamp, GST_BUFFER_TIMESTAMP (buffer));
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fail_unless_equals_uint64 (offset, GST_BUFFER_OFFSET (buffer));
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duration = GST_BUFFER_DURATION (buffer);
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offset_end = GST_BUFFER_OFFSET_END (buffer);
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timestamp += duration;
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offset = offset_end;
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gst_buffer_unref (buffer);
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}
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g_list_free (buffers);
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buffers = NULL;
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}
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/* this tests that the output is a perfect stream if the input is */
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static void
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test_perfect_stream_instance (int inrate, int outrate, int samples,
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int numbuffers)
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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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guint64 offset = 0;
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int i, j;
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GstMapInfo map;
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gint16 *p;
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audioresample =
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setup_audioresample (2, 0x3, inrate, outrate, GST_AUDIO_NE (S16));
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caps = gst_pad_get_current_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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for (j = 1; j <= numbuffers; ++j) {
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inbuffer = gst_buffer_new_and_alloc (samples * 4);
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GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (samples, inrate);
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GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1);
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GST_BUFFER_OFFSET (inbuffer) = offset;
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offset += samples;
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GST_BUFFER_OFFSET_END (inbuffer) = offset;
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gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
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p = (gint16 *) map.data;
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/* create a 16 bit signed ramp */
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for (i = 0; i < samples; ++i) {
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*p = -32767 + i * (65535 / samples);
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++p;
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*p = -32767 + i * (65535 / samples);
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++p;
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}
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gst_buffer_unmap (inbuffer, &map);
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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), j);
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}
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/* FIXME: we should make audioresample handle eos by flushing out the last
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* samples, which will give us one more, small, buffer */
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fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
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ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
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fail_unless_perfect_stream ();
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/* cleanup */
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gst_caps_unref (caps);
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cleanup_audioresample (audioresample);
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}
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/* make sure that outgoing buffers are contiguous in timestamp/duration and
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* offset/offsetend
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*/
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GST_START_TEST (test_perfect_stream)
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{
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/* integral scalings */
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test_perfect_stream_instance (48000, 24000, 500, 20);
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test_perfect_stream_instance (48000, 12000, 500, 20);
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test_perfect_stream_instance (12000, 24000, 500, 20);
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test_perfect_stream_instance (12000, 48000, 500, 20);
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/* non-integral scalings */
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test_perfect_stream_instance (44100, 8000, 500, 20);
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test_perfect_stream_instance (8000, 44100, 500, 20);
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/* wacky scalings */
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test_perfect_stream_instance (12345, 54321, 500, 20);
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test_perfect_stream_instance (101, 99, 500, 20);
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}
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GST_END_TEST;
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/* this tests that the output is a correct discontinuous stream
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* if the input is; ie input drops in time come out the same way */
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static void
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test_discont_stream_instance (int inrate, int outrate, int samples,
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int numbuffers)
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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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GstClockTime ints;
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int i, j;
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GstMapInfo map;
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gint16 *p;
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GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d",
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inrate, outrate, samples, numbuffers);
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audioresample =
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setup_audioresample (2, 3, inrate, outrate, GST_AUDIO_NE (S16));
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caps = gst_pad_get_current_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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for (j = 1; j <= numbuffers; ++j) {
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inbuffer = gst_buffer_new_and_alloc (samples * 4);
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GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate;
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/* "drop" half the buffers */
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ints = GST_BUFFER_DURATION (inbuffer) * 2 * (j - 1);
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GST_BUFFER_TIMESTAMP (inbuffer) = ints;
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GST_BUFFER_OFFSET (inbuffer) = (j - 1) * 2 * samples;
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GST_BUFFER_OFFSET_END (inbuffer) = j * 2 * samples + samples;
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gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
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p = (gint16 *) map.data;
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/* create a 16 bit signed ramp */
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for (i = 0; i < samples; ++i) {
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*p = -32767 + i * (65535 / samples);
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++p;
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*p = -32767 + i * (65535 / samples);
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++p;
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}
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gst_buffer_unmap (inbuffer, &map);
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GST_DEBUG ("Sending Buffer time:%" G_GUINT64_FORMAT " duration:%"
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G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
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G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (inbuffer),
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GST_BUFFER_DURATION (inbuffer), GST_BUFFER_IS_DISCONT (inbuffer),
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GST_BUFFER_OFFSET (inbuffer), GST_BUFFER_OFFSET_END (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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/* check if the timestamp of the pushed buffer matches the incoming one */
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outbuffer = g_list_nth_data (buffers, g_list_length (buffers) - 1);
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fail_if (outbuffer == NULL);
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fail_unless_equals_uint64 (ints, GST_BUFFER_TIMESTAMP (outbuffer));
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GST_DEBUG ("Got Buffer time:%" G_GUINT64_FORMAT " duration:%"
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G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
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G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (outbuffer),
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GST_BUFFER_DURATION (outbuffer), GST_BUFFER_IS_DISCONT (outbuffer),
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GST_BUFFER_OFFSET (outbuffer), GST_BUFFER_OFFSET_END (outbuffer));
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if (j > 1) {
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fail_unless (GST_BUFFER_IS_DISCONT (outbuffer),
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"expected discont for buffer #%d", j);
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}
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}
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/* cleanup */
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gst_caps_unref (caps);
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cleanup_audioresample (audioresample);
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}
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GST_START_TEST (test_discont_stream)
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{
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/* integral scalings */
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test_discont_stream_instance (48000, 24000, 5000, 20);
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test_discont_stream_instance (48000, 12000, 5000, 20);
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test_discont_stream_instance (12000, 24000, 5000, 20);
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test_discont_stream_instance (12000, 48000, 5000, 20);
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/* non-integral scalings */
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test_discont_stream_instance (44100, 8000, 5000, 20);
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test_discont_stream_instance (8000, 44100, 5000, 20);
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/* wacky scalings */
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test_discont_stream_instance (12345, 54321, 5000, 20);
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test_discont_stream_instance (101, 99, 5000, 20);
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}
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GST_END_TEST;
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GST_START_TEST (test_reuse)
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{
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GstElement *audioresample;
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GstEvent *newseg;
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GstBuffer *inbuffer;
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GstCaps *caps;
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GstSegment segment;
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audioresample = setup_audioresample (1, 0, 9343, 48000, GST_AUDIO_NE (S16));
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caps = gst_pad_get_current_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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gst_segment_init (&segment, GST_FORMAT_TIME);
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newseg = gst_event_new_segment (&segment);
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fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
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inbuffer = gst_buffer_new_and_alloc (9343 * 4);
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gst_buffer_memset (inbuffer, 0, 0, 9343 * 4);
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GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
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GST_BUFFER_TIMESTAMP (inbuffer) = 0;
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GST_BUFFER_OFFSET (inbuffer) = 0;
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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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/* now reset and try again ... */
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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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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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newseg = gst_event_new_segment (&segment);
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fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
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inbuffer = gst_buffer_new_and_alloc (9343 * 4);
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gst_buffer_memset (inbuffer, 0, 0, 9343 * 4);
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GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
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GST_BUFFER_TIMESTAMP (inbuffer) = 0;
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GST_BUFFER_OFFSET (inbuffer) = 0;
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fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
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/* ... it also ends up being collected on the global buffer list. If we
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* now have more than 2 buffers, then audioresample probably didn't clean
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* up its internal buffer properly and tried to push the remaining samples
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* when it got the second NEWSEGMENT event */
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fail_unless_equals_int (g_list_length (buffers), 2);
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cleanup_audioresample (audioresample);
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gst_caps_unref (caps);
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}
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GST_END_TEST;
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GST_START_TEST (test_shutdown)
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{
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GstElement *pipeline, *src, *cf1, *ar, *cf2, *sink;
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GstCaps *caps;
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guint i;
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/* create pipeline, force audioresample to actually resample */
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pipeline = gst_pipeline_new (NULL);
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src = gst_check_setup_element ("audiotestsrc");
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cf1 = gst_check_setup_element ("capsfilter");
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ar = gst_check_setup_element ("audioresample");
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cf2 = gst_check_setup_element ("capsfilter");
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g_object_set (cf2, "name", "capsfilter2", NULL);
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sink = gst_check_setup_element ("fakesink");
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caps = gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, 11025, NULL);
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g_object_set (cf1, "caps", caps, NULL);
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gst_caps_unref (caps);
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caps = gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, 48000, NULL);
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g_object_set (cf2, "caps", caps, NULL);
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gst_caps_unref (caps);
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/* don't want to sync against the clock, the more throughput the better */
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g_object_set (src, "is-live", FALSE, NULL);
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g_object_set (sink, "sync", FALSE, NULL);
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gst_bin_add_many (GST_BIN (pipeline), src, cf1, ar, cf2, sink, NULL);
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fail_if (!gst_element_link_many (src, cf1, ar, cf2, sink, NULL));
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/* now, wait until pipeline is running and then shut it down again; repeat */
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for (i = 0; i < 20; ++i) {
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gst_element_set_state (pipeline, GST_STATE_PAUSED);
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gst_element_get_state (pipeline, NULL, NULL, -1);
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gst_element_set_state (pipeline, GST_STATE_PLAYING);
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g_usleep (100);
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gst_element_set_state (pipeline, GST_STATE_NULL);
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}
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gst_object_unref (pipeline);
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}
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GST_END_TEST;
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#if 0
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static GstFlowReturn
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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);
|