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fc8ce00673
Original commit message from CVS: * check/elements/audioconvert.c: * docs/libs/tmpl/gstaudio.sgml: * docs/libs/tmpl/gstcolorbalance.sgml: * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event), (vorbis_handle_identification_packet), (vorbis_handle_data_packet): * ext/vorbis/vorbisenc.c: (raw_caps_factory): * gst-libs/gst/audio/audio.c: (gst_audio_structure_set_int): * gst-libs/gst/audio/audio.h: * gst/audioconvert/audioconvert.h: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_parse_caps): * gst/volume/gstvolume.c: Bye bye buffer-frames.
315 lines
9.9 KiB
C
315 lines
9.9 KiB
C
/* GStreamer
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*
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* unit test for audioconvert
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*
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* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot 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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, 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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GList *buffers = NULL;
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gboolean have_eos = FALSE;
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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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GstPad *mysrcpad, *mysinkpad;
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#define CONVERT_CAPS_TEMPLATE_STRING \
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"audio/x-raw-float, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) 32;" \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) 32, " \
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"depth = (int) [ 1, 32 ], " \
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"signed = (boolean) { true, false }; " \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) 24, " \
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"depth = (int) [ 1, 24 ], " \
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"signed = (boolean) { true, false }; " \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) 16, " \
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"depth = (int) [ 1, 16 ], " \
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"signed = (boolean) { true, false }; " \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) 8, " \
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"depth = (int) [ 1, 8 ], " \
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"signed = (boolean) { true, false } "
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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 (CONVERT_CAPS_TEMPLATE_STRING)
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);
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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 (CONVERT_CAPS_TEMPLATE_STRING)
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);
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/* takes over reference for outcaps */
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GstElement *
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setup_audioconvert (GstCaps * outcaps)
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{
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GstElement *audioconvert;
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GST_DEBUG ("setup_audioconvert with caps %" GST_PTR_FORMAT, outcaps);
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audioconvert = gst_check_setup_element ("audioconvert");
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mysrcpad = gst_check_setup_src_pad (audioconvert, &srctemplate, NULL);
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mysinkpad = gst_check_setup_sink_pad (audioconvert, &sinktemplate, NULL);
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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_pad_set_caps (mysinkpad, outcaps);
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gst_caps_unref (outcaps);
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outcaps = gst_pad_get_negotiated_caps (mysinkpad);
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fail_unless (gst_caps_is_fixed (outcaps));
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gst_caps_unref (outcaps);
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gst_pad_set_active (mysrcpad, TRUE);
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gst_pad_set_active (mysinkpad, TRUE);
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return audioconvert;
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}
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void
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cleanup_audioconvert (GstElement * audioconvert)
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{
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GST_DEBUG ("cleanup_audioconvert");
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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 (audioconvert);
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gst_check_teardown_sink_pad (audioconvert);
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gst_check_teardown_element (audioconvert);
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}
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/* returns a newly allocated caps */
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static GstCaps *
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get_int_caps (guint channels, gchar * endianness, guint width,
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guint depth, gboolean signedness)
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{
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GstCaps *caps;
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gchar *string;
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string = g_strdup_printf ("audio/x-raw-int, "
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"rate = (int) 44100, "
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"channels = (int) %d, "
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"endianness = (int) %s, "
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"width = (int) %d, "
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"depth = (int) %d, "
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"signed = (boolean) %s ",
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channels, endianness, width, depth, signedness ? "true" : "false");
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GST_DEBUG ("creating caps from %s", string);
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caps = gst_caps_from_string (string);
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g_free (string);
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fail_unless (caps != NULL);
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GST_DEBUG ("returning caps %p", caps);
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return caps;
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}
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/* eats the refs to the caps */
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static void
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verify_convert (void *in, int inlength,
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GstCaps * incaps, void *out, int outlength, GstCaps * outcaps)
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{
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GstBuffer *inbuffer, *outbuffer;
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GstElement *audioconvert;
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GST_DEBUG ("incaps: %" GST_PTR_FORMAT, incaps);
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GST_DEBUG ("outcaps: %" GST_PTR_FORMAT, outcaps);
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ASSERT_CAPS_REFCOUNT (incaps, "incaps", 1);
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ASSERT_CAPS_REFCOUNT (outcaps, "outcaps", 1);
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audioconvert = setup_audioconvert (outcaps);
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ASSERT_CAPS_REFCOUNT (outcaps, "outcaps", 1);
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fail_unless (gst_element_set_state (audioconvert,
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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_DEBUG ("Creating buffer of %d bytes", inlength);
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inbuffer = gst_buffer_new_and_alloc (inlength);
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memcpy (GST_BUFFER_DATA (inbuffer), in, inlength);
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gst_buffer_set_caps (inbuffer, incaps);
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ASSERT_CAPS_REFCOUNT (incaps, "incaps", 2);
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ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
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/* pushing gives away my reference ... */
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GST_DEBUG ("push it");
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fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
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GST_DEBUG ("pushed it");
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/* ... and puts a new buffer on the global list */
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fail_unless (g_list_length (buffers) == 1);
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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_equals_int (GST_BUFFER_SIZE (outbuffer), outlength);
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fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, outlength) == 0);
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buffers = g_list_remove (buffers, outbuffer);
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gst_buffer_unref (outbuffer);
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fail_unless (gst_element_set_state (audioconvert,
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GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to null");
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/* cleanup */
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GST_DEBUG ("cleanup audioconvert");
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cleanup_audioconvert (audioconvert);
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GST_DEBUG ("cleanup, unref incaps");
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ASSERT_CAPS_REFCOUNT (incaps, "incaps", 1);
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gst_caps_unref (incaps);
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}
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#define RUN_CONVERSION(inarray, in_get_caps, outarray, out_get_caps) \
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verify_convert (inarray, sizeof (inarray), \
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in_get_caps, outarray, sizeof (outarray), out_get_caps)
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GST_START_TEST (test_int16)
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{
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/* stereo to mono */
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{
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gint16 in[] = { 16384, -256, 1024, 1024 };
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gint16 out[] = { 8064, 1024 };
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RUN_CONVERSION (in, get_int_caps (2, "BYTE_ORDER", 16, 16, TRUE),
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out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE));
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}
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/* mono to stereo */
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{
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gint16 in[] = { 512, 1024 };
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gint16 out[] = { 512, 512, 1024, 1024 };
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RUN_CONVERSION (in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
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out, get_int_caps (2, "BYTE_ORDER", 16, 16, TRUE));
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}
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/* signed -> unsigned */
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{
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gint16 in[] = { 0, -32767, 32767, -32768 };
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guint16 out[] = { 32768, 1, 65535, 0 };
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RUN_CONVERSION (in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
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out, get_int_caps (1, "BYTE_ORDER", 16, 16, FALSE));
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RUN_CONVERSION (out, get_int_caps (1, "BYTE_ORDER", 16, 16, FALSE),
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in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE));
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}
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}
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GST_END_TEST;
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GST_START_TEST (test_int_conversion)
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{
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/* 8 <-> 16 signed */
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/* NOTE: if audioconvert was doing dithering we'd have a problem */
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{
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gint8 in[] = { 0, 1, 2, 127, -127 };
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gint16 out[] = { 0, 256, 512, 32512, -32512 };
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RUN_CONVERSION (in, get_int_caps (1, "BYTE_ORDER", 8, 8, TRUE),
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out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE)
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);
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RUN_CONVERSION (out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
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in, get_int_caps (1, "BYTE_ORDER", 8, 8, TRUE)
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);
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}
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/* 16 -> 8 signed */
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{
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gint16 in[] = { 0, 255, 256, 257 };
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gint8 out[] = { 0, 0, 1, 1 };
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RUN_CONVERSION (in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
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out, get_int_caps (1, "BYTE_ORDER", 8, 8, TRUE)
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);
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}
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/* 8 unsigned <-> 16 signed */
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/* NOTE: if audioconvert was doing dithering we'd have a problem */
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{
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guint8 in[] = { 128, 129, 130, 255, 1 };
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gint16 out[] = { 0, 256, 512, 32512, -32512 };
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GstCaps *incaps, *outcaps;
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/* exploded for easier valgrinding */
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incaps = get_int_caps (1, "BYTE_ORDER", 8, 8, FALSE);
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outcaps = get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE);
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GST_DEBUG ("incaps: %" GST_PTR_FORMAT, incaps);
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GST_DEBUG ("outcaps: %" GST_PTR_FORMAT, outcaps);
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RUN_CONVERSION (in, incaps, out, outcaps);
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RUN_CONVERSION (out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
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in, get_int_caps (1, "BYTE_ORDER", 8, 8, FALSE)
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);
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}
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/* 8 <-> 24 signed */
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/* NOTE: if audioconvert was doing dithering we'd have a problem */
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{
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gint8 in[] = { 0, 1, 127 };
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guint8 out[] = { 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x7f };
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RUN_CONVERSION (in, get_int_caps (1, "BYTE_ORDER", 8, 8, TRUE),
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out, get_int_caps (1, "BYTE_ORDER", 24, 24, TRUE)
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);
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RUN_CONVERSION (out, get_int_caps (1, "BYTE_ORDER", 24, 24, TRUE),
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in, get_int_caps (1, "BYTE_ORDER", 8, 8, TRUE)
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);
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}
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}
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GST_END_TEST;
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Suite *
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audioconvert_suite (void)
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{
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Suite *s = suite_create ("audioconvert");
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TCase *tc_chain = tcase_create ("general");
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suite_add_tcase (s, tc_chain);
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tcase_add_test (tc_chain, test_int16);
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//tcase_add_test (tc_chain, test_int_conversion);
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return s;
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}
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int
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main (int argc, char **argv)
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{
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int nf;
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Suite *s = audioconvert_suite ();
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SRunner *sr = srunner_create (s);
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gst_check_init (&argc, &argv);
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srunner_run_all (sr, CK_NORMAL);
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nf = srunner_ntests_failed (sr);
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srunner_free (sr);
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return nf;
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}
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