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dbb857b93b
Original commit message from CVS: * gst/audioconvert/Makefile.am: * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index), (check_default), (audio_convert_prepare_context), (audio_convert_clean_context), (audio_convert_convert): * gst/audioconvert/audioconvert.h: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dithering_get_type), (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init), (gst_audio_convert_init), (gst_audio_convert_set_caps), (gst_audio_convert_set_property), (gst_audio_convert_get_property): * gst/audioconvert/gstaudioconvert.h: * gst/audioconvert/gstaudioquantize.c: (gst_audio_quantize_setup_noise_shaping), (gst_audio_quantize_free_noise_shaping), (gst_audio_quantize_setup_dither), (gst_audio_quantize_free_dither), (gst_audio_quantize_setup_quantize_func), (gst_audio_quantize_setup), (gst_audio_quantize_free): * gst/audioconvert/gstaudioquantize.h: Implement dithering and noise shaping in audioconvert. By default now TPDF dithering (and no noise shaping) will be used when converting from a higher bit depth to 20 bit depth or smaller, otherwise everything will be as it is now. For the last audioconvert in a pipeline it would make sense to use some kind of noise shaping, enabling it by default for all conversions would give undesired results though. Fixes #360246. * tests/check/elements/audioconvert.c: (setup_audioconvert), (GST_START_TEST): Adjust unit test for the new audioconvert.
867 lines
29 KiB
C
867 lines
29 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/floatcast/floatcast.h>
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#include <gst/check/gstcheck.h>
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#include <gst/audio/multichannel.h>
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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) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) { 32, 64 };" \
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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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g_object_set (G_OBJECT (audioconvert), "dithering", 0, NULL);
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g_object_set (G_OBJECT (audioconvert), "noise-shaping", 0, NULL);
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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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/* returns a newly allocated caps */
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static GstCaps *
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get_float_caps (guint channels, gchar * endianness, guint width)
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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-float, "
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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 ", channels, endianness, width);
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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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/* Copied from vorbis; the particular values used don't matter */
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static GstAudioChannelPosition channelpositions[][6] = {
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{ /* Mono */
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GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
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{ /* Stereo */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
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{ /* Stereo + Centre */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
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{ /* Quadraphonic */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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},
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{ /* Stereo + Centre + rear stereo */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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},
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{ /* Full 5.1 Surround */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_LFE,
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}
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};
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/* these are a bunch of random positions, they are mostly just
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* different from the ones above, don't use elsewhere */
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static GstAudioChannelPosition mixed_up_positions[][6] = {
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{
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GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
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{
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT},
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{
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT},
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{
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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},
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{
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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},
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{
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_LFE,
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}
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};
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static void
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set_channel_positions (GstCaps * caps, int channels,
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GstAudioChannelPosition * channelpositions)
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{
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GValue chanpos = { 0 };
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GValue pos = { 0 };
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GstStructure *structure = gst_caps_get_structure (caps, 0);
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int c;
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g_value_init (&chanpos, GST_TYPE_ARRAY);
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g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
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for (c = 0; c < channels; c++) {
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g_value_set_enum (&pos, channelpositions[c]);
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gst_value_array_append_value (&chanpos, &pos);
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}
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g_value_unset (&pos);
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gst_structure_set_value (structure, "channel-positions", &chanpos);
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g_value_unset (&chanpos);
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}
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/* For channels > 2, caps have to have channel positions. This adds some simple
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* ones. Only implemented for channels between 1 and 6.
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*/
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static GstCaps *
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get_float_mc_caps (guint channels, gchar * endianness, guint width,
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gboolean mixed_up_layout)
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{
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GstCaps *caps = get_float_caps (channels, endianness, width);
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if (channels <= 6) {
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if (mixed_up_layout)
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set_channel_positions (caps, channels, mixed_up_positions[channels - 1]);
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else
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set_channel_positions (caps, channels, channelpositions[channels - 1]);
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}
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return caps;
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}
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static GstCaps *
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get_int_mc_caps (guint channels, gchar * endianness, guint width,
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guint depth, gboolean signedness, gboolean mixed_up_layout)
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{
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GstCaps *caps = get_int_caps (channels, endianness, width, depth, signedness);
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if (channels <= 6) {
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if (mixed_up_layout)
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set_channel_positions (caps, channels, mixed_up_positions[channels - 1]);
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else
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set_channel_positions (caps, channels, channelpositions[channels - 1]);
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}
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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 (const gchar * which, 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 ("verifying conversion %s", which);
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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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if (memcmp (GST_BUFFER_DATA (outbuffer), out, outlength) != 0) {
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g_print ("\nInput data:\n");
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gst_util_dump_mem (in, inlength);
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g_print ("\nConverted data:\n");
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gst_util_dump_mem (GST_BUFFER_DATA (outbuffer), outlength);
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g_print ("\nExpected data:\n");
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gst_util_dump_mem (out, outlength);
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}
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fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, outlength) == 0,
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"failed converting %s", which);
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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(which, inarray, in_get_caps, outarray, out_get_caps) \
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verify_convert (which, 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 ("int16 stereo to mono",
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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 ("int16 mono to stereo",
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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 ("int16 signed to unsigned",
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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 ("int16 unsigned to signed",
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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_float32)
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{
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/* stereo to mono */
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{
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gfloat in[] = { 0.6, -0.0078125, 0.03125, 0.03125 };
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gfloat out[] = { 0.29609375, 0.03125 };
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RUN_CONVERSION ("float32 stereo to mono",
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in, get_float_caps (2, "BYTE_ORDER", 32),
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out, get_float_caps (1, "BYTE_ORDER", 32));
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}
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/* mono to stereo */
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{
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gfloat in[] = { 0.015625, 0.03125 };
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gfloat out[] = { 0.015625, 0.015625, 0.03125, 0.03125 };
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RUN_CONVERSION ("float32 mono to stereo",
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in, get_float_caps (1, "BYTE_ORDER", 32),
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out, get_float_caps (2, "BYTE_ORDER", 32));
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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 */
|
|
/* NOTE: if audioconvert was doing dithering we'd have a problem */
|
|
{
|
|
gint8 in[] = { 0, 1, 2, 127, -127 };
|
|
gint16 out[] = { 0, 256, 512, 32512, -32512 };
|
|
|
|
RUN_CONVERSION ("int 8bit to 16bit signed",
|
|
in, get_int_caps (1, "BYTE_ORDER", 8, 8, TRUE),
|
|
out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE)
|
|
);
|
|
RUN_CONVERSION ("int 16bit signed to 8bit",
|
|
out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
|
|
in, get_int_caps (1, "BYTE_ORDER", 8, 8, TRUE)
|
|
);
|
|
}
|
|
/* 16 -> 8 signed */
|
|
{
|
|
gint16 in[] = { 0, 127, 128, 256, 256 + 127, 256 + 128 };
|
|
gint8 out[] = { 0, 0, 1, 1, 1, 2 };
|
|
|
|
RUN_CONVERSION ("16 bit to 8 signed",
|
|
in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
|
|
out, get_int_caps (1, "BYTE_ORDER", 8, 8, TRUE)
|
|
);
|
|
}
|
|
/* 8 unsigned <-> 16 signed */
|
|
/* NOTE: if audioconvert was doing dithering we'd have a problem */
|
|
{
|
|
guint8 in[] = { 128, 129, 130, 255, 1 };
|
|
gint16 out[] = { 0, 256, 512, 32512, -32512 };
|
|
GstCaps *incaps, *outcaps;
|
|
|
|
/* exploded for easier valgrinding */
|
|
incaps = get_int_caps (1, "BYTE_ORDER", 8, 8, FALSE);
|
|
outcaps = get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE);
|
|
GST_DEBUG ("incaps: %" GST_PTR_FORMAT, incaps);
|
|
GST_DEBUG ("outcaps: %" GST_PTR_FORMAT, outcaps);
|
|
RUN_CONVERSION ("8 unsigned to 16 signed", in, incaps, out, outcaps);
|
|
RUN_CONVERSION ("16 signed to 8 unsigned", out, get_int_caps (1,
|
|
"BYTE_ORDER", 16, 16, TRUE), in, get_int_caps (1, "BYTE_ORDER", 8,
|
|
8, FALSE)
|
|
);
|
|
}
|
|
/* 8 <-> 24 signed */
|
|
/* NOTE: if audioconvert was doing dithering we'd have a problem */
|
|
{
|
|
gint8 in[] = { 0, 1, 127 };
|
|
guint8 out[] = { 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x7f };
|
|
/* out has the bytes in little-endian, so that's how they should be
|
|
* interpreted during conversion */
|
|
|
|
RUN_CONVERSION ("8 to 24 signed", in, get_int_caps (1, "BYTE_ORDER", 8, 8,
|
|
TRUE), out, get_int_caps (1, "LITTLE_ENDIAN", 24, 24, TRUE)
|
|
);
|
|
RUN_CONVERSION ("24 signed to 8", out, get_int_caps (1, "LITTLE_ENDIAN", 24,
|
|
24, TRUE), in, get_int_caps (1, "BYTE_ORDER", 8, 8, TRUE)
|
|
);
|
|
}
|
|
|
|
/* 16 bit signed <-> unsigned */
|
|
{
|
|
gint16 in[] = { 0, 128, -128 };
|
|
guint16 out[] = { 32768, 32896, 32640 };
|
|
RUN_CONVERSION ("16 signed to 16 unsigned",
|
|
in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
|
|
out, get_int_caps (1, "BYTE_ORDER", 16, 16, FALSE)
|
|
);
|
|
RUN_CONVERSION ("16 unsigned to 16 signed",
|
|
out, get_int_caps (1, "BYTE_ORDER", 16, 16, FALSE),
|
|
in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE)
|
|
);
|
|
}
|
|
|
|
/* 16 bit signed <-> 8 in 16 bit signed */
|
|
/* NOTE: if audioconvert was doing dithering we'd have a problem */
|
|
{
|
|
gint16 in[] = { 0, 64 << 8, -64 << 8 };
|
|
gint16 out[] = { 0, 64, -64 };
|
|
RUN_CONVERSION ("16 signed to 8 in 16 signed",
|
|
in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
|
|
out, get_int_caps (1, "BYTE_ORDER", 16, 8, TRUE)
|
|
);
|
|
RUN_CONVERSION ("8 in 16 signed to 16 signed",
|
|
out, get_int_caps (1, "BYTE_ORDER", 16, 8, TRUE),
|
|
in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE)
|
|
);
|
|
}
|
|
|
|
/* 16 bit unsigned <-> 8 in 16 bit unsigned */
|
|
/* NOTE: if audioconvert was doing dithering we'd have a problem */
|
|
{
|
|
guint16 in[] = { 1 << 15, (1 << 15) - (64 << 8), (1 << 15) + (64 << 8) };
|
|
guint16 out[] = { 1 << 7, (1 << 7) - 64, (1 << 7) + 64 };
|
|
RUN_CONVERSION ("16 unsigned to 8 in 16 unsigned",
|
|
in, get_int_caps (1, "BYTE_ORDER", 16, 16, FALSE),
|
|
out, get_int_caps (1, "BYTE_ORDER", 16, 8, FALSE)
|
|
);
|
|
RUN_CONVERSION ("8 in 16 unsigned to 16 unsigned",
|
|
out, get_int_caps (1, "BYTE_ORDER", 16, 8, FALSE),
|
|
in, get_int_caps (1, "BYTE_ORDER", 16, 16, FALSE)
|
|
);
|
|
}
|
|
|
|
/* 32 bit signed -> 16 bit signed for rounding check */
|
|
/* NOTE: if audioconvert was doing dithering we'd have a problem */
|
|
{
|
|
gint32 in[] = { 0, G_MININT32, G_MAXINT32,
|
|
(32 << 16), (32 << 16) + (1 << 15), (32 << 16) - (1 << 15),
|
|
(32 << 16) + (2 << 15), (32 << 16) - (2 << 15),
|
|
(-32 << 16) + (1 << 15), (-32 << 16) - (1 << 15),
|
|
(-32 << 16) + (2 << 15), (-32 << 16) - (2 << 15),
|
|
(-32 << 16)
|
|
};
|
|
gint16 out[] = { 0, G_MININT16, G_MAXINT16,
|
|
32, 33, 32,
|
|
33, 31,
|
|
-31, -32,
|
|
-31, -33,
|
|
-32
|
|
};
|
|
RUN_CONVERSION ("32 signed to 16 signed for rounding",
|
|
in, get_int_caps (1, "BYTE_ORDER", 32, 32, TRUE),
|
|
out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE)
|
|
);
|
|
}
|
|
|
|
/* 32 bit signed -> 16 bit unsigned for rounding check */
|
|
/* NOTE: if audioconvert was doing dithering we'd have a problem */
|
|
{
|
|
gint32 in[] = { 0, G_MININT32, G_MAXINT32,
|
|
(32 << 16), (32 << 16) + (1 << 15), (32 << 16) - (1 << 15),
|
|
(32 << 16) + (2 << 15), (32 << 16) - (2 << 15),
|
|
(-32 << 16) + (1 << 15), (-32 << 16) - (1 << 15),
|
|
(-32 << 16) + (2 << 15), (-32 << 16) - (2 << 15),
|
|
(-32 << 16)
|
|
};
|
|
guint16 out[] = { (1 << 15), 0, G_MAXUINT16,
|
|
(1 << 15) + 32, (1 << 15) + 33, (1 << 15) + 32,
|
|
(1 << 15) + 33, (1 << 15) + 31,
|
|
(1 << 15) - 31, (1 << 15) - 32,
|
|
(1 << 15) - 31, (1 << 15) - 33,
|
|
(1 << 15) - 32
|
|
};
|
|
RUN_CONVERSION ("32 signed to 16 unsigned for rounding",
|
|
in, get_int_caps (1, "BYTE_ORDER", 32, 32, TRUE),
|
|
out, get_int_caps (1, "BYTE_ORDER", 16, 16, FALSE)
|
|
);
|
|
}
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_float_conversion)
|
|
{
|
|
/* 32 float <-> 16 signed */
|
|
/* NOTE: if audioconvert was doing dithering we'd have a problem */
|
|
{
|
|
gfloat in_le[] =
|
|
{ GFLOAT_TO_LE (0.0), GFLOAT_TO_LE (1.0), GFLOAT_TO_LE (-1.0),
|
|
GFLOAT_TO_LE (0.5), GFLOAT_TO_LE (-0.5), GFLOAT_TO_LE (1.1),
|
|
GFLOAT_TO_LE (-1.1)
|
|
};
|
|
gfloat in_be[] =
|
|
{ GFLOAT_TO_BE (0.0), GFLOAT_TO_BE (1.0), GFLOAT_TO_BE (-1.0),
|
|
GFLOAT_TO_BE (0.5), GFLOAT_TO_BE (-0.5), GFLOAT_TO_BE (1.1),
|
|
GFLOAT_TO_BE (-1.1)
|
|
};
|
|
gint16 out[] = { 0, 32767, -32768, 16384, -16384, 32767, -32768 };
|
|
|
|
/* only one direction conversion, the other direction does
|
|
* not produce exactly the same as the input due to floating
|
|
* point rounding errors etc. */
|
|
RUN_CONVERSION ("32 float le to 16 signed",
|
|
in_le, get_float_caps (1, "LITTLE_ENDIAN", 32),
|
|
out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE));
|
|
RUN_CONVERSION ("32 float be to 16 signed",
|
|
in_be, get_float_caps (1, "BIG_ENDIAN", 32),
|
|
out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE));
|
|
}
|
|
|
|
{
|
|
gint16 in[] = { 0, -32768, 16384, -16384 };
|
|
gfloat out[] = { 0.0, -1.0, 0.5, -0.5 };
|
|
|
|
RUN_CONVERSION ("16 signed to 32 float",
|
|
in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
|
|
out, get_float_caps (1, "BYTE_ORDER", 32));
|
|
}
|
|
|
|
/* 64 float <-> 16 signed */
|
|
/* NOTE: if audioconvert was doing dithering we'd have a problem */
|
|
{
|
|
gdouble in_le[] =
|
|
{ GDOUBLE_TO_LE (0.0), GDOUBLE_TO_LE (1.0), GDOUBLE_TO_LE (-1.0),
|
|
GDOUBLE_TO_LE (0.5), GDOUBLE_TO_LE (-0.5), GDOUBLE_TO_LE (1.1),
|
|
GDOUBLE_TO_LE (-1.1)
|
|
};
|
|
gdouble in_be[] =
|
|
{ GDOUBLE_TO_BE (0.0), GDOUBLE_TO_BE (1.0), GDOUBLE_TO_BE (-1.0),
|
|
GDOUBLE_TO_BE (0.5), GDOUBLE_TO_BE (-0.5), GDOUBLE_TO_BE (1.1),
|
|
GDOUBLE_TO_BE (-1.1)
|
|
};
|
|
gint16 out[] = { 0, 32767, -32768, 16384, -16384, 32767, -32768 };
|
|
|
|
/* only one direction conversion, the other direction does
|
|
* not produce exactly the same as the input due to floating
|
|
* point rounding errors etc. */
|
|
RUN_CONVERSION ("64 float LE to 16 signed",
|
|
in_le, get_float_caps (1, "LITTLE_ENDIAN", 64),
|
|
out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE));
|
|
RUN_CONVERSION ("64 float BE to 16 signed",
|
|
in_be, get_float_caps (1, "BIG_ENDIAN", 64),
|
|
out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE));
|
|
}
|
|
{
|
|
gint16 in[] = { 0, -32768, 16384, -16384 };
|
|
gdouble out[] = { 0.0,
|
|
(gdouble) (-32768L << 16) / 2147483647.0, /* ~ -1.0 */
|
|
(gdouble) (16384L << 16) / 2147483647.0, /* ~ 0.5 */
|
|
(gdouble) (-16384L << 16) / 2147483647.0, /* ~ -0.5 */
|
|
};
|
|
|
|
RUN_CONVERSION ("16 signed to 64 float",
|
|
in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
|
|
out, get_float_caps (1, "BYTE_ORDER", 64));
|
|
}
|
|
{
|
|
gint32 in[] = { 0, (-1L << 31), (1L << 30), (-1L << 30) };
|
|
gdouble out[] = { 0.0,
|
|
(gdouble) (-1L << 31) / 2147483647.0, /* ~ -1.0 */
|
|
(gdouble) (1L << 30) / 2147483647.0, /* ~ 0.5 */
|
|
(gdouble) (-1L << 30) / 2147483647.0, /* ~ -0.5 */
|
|
};
|
|
|
|
RUN_CONVERSION ("32 signed to 64 float",
|
|
in, get_int_caps (1, "BYTE_ORDER", 32, 32, TRUE),
|
|
out, get_float_caps (1, "BYTE_ORDER", 64));
|
|
}
|
|
|
|
/* 64-bit float <-> 32-bit float */
|
|
{
|
|
gdouble in[] = { 0.0, 1.0, -1.0, 0.5, -0.5 };
|
|
gfloat out[] = { 0.0, 1.0, -1.0, 0.5, -0.5 };
|
|
|
|
RUN_CONVERSION ("64 float to 32 float",
|
|
in, get_float_caps (1, "BYTE_ORDER", 64),
|
|
out, get_float_caps (1, "BYTE_ORDER", 32));
|
|
|
|
RUN_CONVERSION ("32 float to 64 float",
|
|
out, get_float_caps (1, "BYTE_ORDER", 32),
|
|
in, get_float_caps (1, "BYTE_ORDER", 64));
|
|
}
|
|
|
|
/* 32-bit float little endian <-> big endian */
|
|
{
|
|
gfloat le[] = { GFLOAT_TO_LE (0.0), GFLOAT_TO_LE (1.0), GFLOAT_TO_LE (-1.0),
|
|
GFLOAT_TO_LE (0.5), GFLOAT_TO_LE (-0.5)
|
|
};
|
|
gfloat be[] = { GFLOAT_TO_BE (0.0), GFLOAT_TO_BE (1.0), GFLOAT_TO_BE (-1.0),
|
|
GFLOAT_TO_BE (0.5), GFLOAT_TO_BE (-0.5)
|
|
};
|
|
|
|
RUN_CONVERSION ("32 float LE to BE",
|
|
le, get_float_caps (1, "LITTLE_ENDIAN", 32),
|
|
be, get_float_caps (1, "BIG_ENDIAN", 32));
|
|
|
|
RUN_CONVERSION ("32 float BE to LE",
|
|
be, get_float_caps (1, "BIG_ENDIAN", 32),
|
|
le, get_float_caps (1, "LITTLE_ENDIAN", 32));
|
|
}
|
|
|
|
/* 64-bit float little endian <-> big endian */
|
|
{
|
|
gdouble le[] =
|
|
{ GDOUBLE_TO_LE (0.0), GDOUBLE_TO_LE (1.0), GDOUBLE_TO_LE (-1.0),
|
|
GDOUBLE_TO_LE (0.5), GDOUBLE_TO_LE (-0.5)
|
|
};
|
|
gdouble be[] =
|
|
{ GDOUBLE_TO_BE (0.0), GDOUBLE_TO_BE (1.0), GDOUBLE_TO_BE (-1.0),
|
|
GDOUBLE_TO_BE (0.5), GDOUBLE_TO_BE (-0.5)
|
|
};
|
|
|
|
RUN_CONVERSION ("64 float LE to BE",
|
|
le, get_float_caps (1, "LITTLE_ENDIAN", 64),
|
|
be, get_float_caps (1, "BIG_ENDIAN", 64));
|
|
|
|
RUN_CONVERSION ("64 float BE to LE",
|
|
be, get_float_caps (1, "BIG_ENDIAN", 64),
|
|
le, get_float_caps (1, "LITTLE_ENDIAN", 64));
|
|
}
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
|
|
GST_START_TEST (test_multichannel_conversion)
|
|
{
|
|
{
|
|
/* Ensure that audioconvert prefers to convert to integer, rather than mix
|
|
* to mono
|
|
*/
|
|
gfloat in[] = { 0.0, 0.0, 0.0, 0.0, 0.0, 0.0 };
|
|
gfloat out[] = { 0.0, 0.0 };
|
|
|
|
/* only one direction conversion, the other direction does
|
|
* not produce exactly the same as the input due to floating
|
|
* point rounding errors etc. */
|
|
RUN_CONVERSION ("3 channels to 1", in, get_float_mc_caps (3,
|
|
"BYTE_ORDER", 32, FALSE), out, get_float_caps (1, "BYTE_ORDER",
|
|
32));
|
|
}
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
|
|
GST_START_TEST (test_channel_remapping)
|
|
{
|
|
/* float */
|
|
{
|
|
gfloat in[] = { 0.0, 1.0, -0.5 };
|
|
gfloat out[] = { -0.5, 1.0, 0.0 };
|
|
GstCaps *in_caps = get_float_mc_caps (3, "BYTE_ORDER", 32, FALSE);
|
|
GstCaps *out_caps = get_float_mc_caps (3, "BYTE_ORDER", 32, TRUE);
|
|
|
|
RUN_CONVERSION ("3 channels layout remapping float", in, in_caps,
|
|
out, out_caps);
|
|
}
|
|
|
|
/* int */
|
|
{
|
|
guint16 in[] = { 0, 65535, 0x9999 };
|
|
guint16 out[] = { 0x9999, 65535, 0 };
|
|
GstCaps *in_caps = get_int_mc_caps (3, "BYTE_ORDER", 16, 16, FALSE, FALSE);
|
|
GstCaps *out_caps = get_int_mc_caps (3, "BYTE_ORDER", 16, 16, FALSE, TRUE);
|
|
|
|
RUN_CONVERSION ("3 channels layout remapping int", in, in_caps,
|
|
out, out_caps);
|
|
}
|
|
|
|
/* TODO: float => int conversion with remapping and vice versa,
|
|
* int => int conversion with remapping */
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_caps_negotiation)
|
|
{
|
|
GstElement *src, *ac1, *ac2, *ac3, *sink;
|
|
GstElement *pipeline;
|
|
GstPad *ac3_src;
|
|
GstCaps *caps1, *caps2;
|
|
|
|
pipeline = gst_pipeline_new ("test");
|
|
|
|
/* create elements */
|
|
src = gst_element_factory_make ("audiotestsrc", "src");
|
|
ac1 = gst_element_factory_make ("audioconvert", "ac1");
|
|
ac2 = gst_element_factory_make ("audioconvert", "ac2");
|
|
ac3 = gst_element_factory_make ("audioconvert", "ac3");
|
|
sink = gst_element_factory_make ("fakesink", "sink");
|
|
ac3_src = gst_element_get_pad (ac3, "src");
|
|
|
|
/* test with 2 audioconvert elements */
|
|
gst_bin_add_many (GST_BIN (pipeline), src, ac1, ac3, sink, NULL);
|
|
gst_element_link_many (src, ac1, ac3, sink, NULL);
|
|
|
|
/* Set to PAUSED and wait for PREROLL */
|
|
fail_if (gst_element_set_state (pipeline, GST_STATE_PAUSED) ==
|
|
GST_STATE_CHANGE_FAILURE, "Failed to set test pipeline to PAUSED");
|
|
fail_if (gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE) !=
|
|
GST_STATE_CHANGE_SUCCESS, "Failed to set test pipeline to PAUSED");
|
|
|
|
caps1 = gst_pad_get_caps (ac3_src);
|
|
fail_if (caps1 == NULL, "gst_pad_get_caps returned NULL");
|
|
GST_DEBUG ("Caps size 1 : %d", gst_caps_get_size (caps1));
|
|
|
|
fail_if (gst_element_set_state (pipeline, GST_STATE_READY) ==
|
|
GST_STATE_CHANGE_FAILURE, "Failed to set test pipeline back to READY");
|
|
fail_if (gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE) !=
|
|
GST_STATE_CHANGE_SUCCESS, "Failed to set test pipeline back to READY");
|
|
|
|
/* test with 3 audioconvert elements */
|
|
gst_element_unlink (ac1, ac3);
|
|
gst_bin_add (GST_BIN (pipeline), ac2);
|
|
gst_element_link_many (ac1, ac2, ac3, NULL);
|
|
|
|
fail_if (gst_element_set_state (pipeline, GST_STATE_PAUSED) ==
|
|
GST_STATE_CHANGE_FAILURE, "Failed to set test pipeline back to PAUSED");
|
|
fail_if (gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE) !=
|
|
GST_STATE_CHANGE_SUCCESS, "Failed to set test pipeline back to PAUSED");
|
|
|
|
caps2 = gst_pad_get_caps (ac3_src);
|
|
|
|
fail_if (caps2 == NULL, "gst_pad_get_caps returned NULL");
|
|
GST_DEBUG ("Caps size 2 : %d", gst_caps_get_size (caps2));
|
|
fail_unless (gst_caps_get_size (caps1) == gst_caps_get_size (caps2));
|
|
|
|
gst_caps_unref (caps1);
|
|
gst_caps_unref (caps2);
|
|
|
|
fail_if (gst_element_set_state (pipeline, GST_STATE_NULL) ==
|
|
GST_STATE_CHANGE_FAILURE, "Failed to set test pipeline back to NULL");
|
|
fail_if (gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE) !=
|
|
GST_STATE_CHANGE_SUCCESS, "Failed to set test pipeline back to NULL");
|
|
|
|
gst_object_unref (ac3_src);
|
|
gst_object_unref (pipeline);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
|
|
Suite *
|
|
audioconvert_suite (void)
|
|
{
|
|
Suite *s = suite_create ("audioconvert");
|
|
TCase *tc_chain = tcase_create ("general");
|
|
|
|
suite_add_tcase (s, tc_chain);
|
|
tcase_add_test (tc_chain, test_int16);
|
|
tcase_add_test (tc_chain, test_float32);
|
|
tcase_add_test (tc_chain, test_int_conversion);
|
|
tcase_add_test (tc_chain, test_float_conversion);
|
|
tcase_add_test (tc_chain, test_multichannel_conversion);
|
|
tcase_add_test (tc_chain, test_channel_remapping);
|
|
tcase_add_test (tc_chain, test_caps_negotiation);
|
|
|
|
return s;
|
|
}
|
|
|
|
int
|
|
main (int argc, char **argv)
|
|
{
|
|
int nf;
|
|
|
|
Suite *s = audioconvert_suite ();
|
|
SRunner *sr = srunner_create (s);
|
|
|
|
gst_check_init (&argc, &argv);
|
|
|
|
srunner_run_all (sr, CK_NORMAL);
|
|
nf = srunner_ntests_failed (sr);
|
|
srunner_free (sr);
|
|
|
|
return nf;
|
|
}
|