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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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dfc82f3466
Copied from videodecoder tests and updated to audio features
912 lines
25 KiB
C
912 lines
25 KiB
C
/* GStreamer
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*
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* Copyright (C) 2014 Samsung Electronics. All rights reserved.
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* Author: Thiago Santos <ts.santos@sisa.samsung.com>
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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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/gst.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/app/app.h>
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static GstPad *mysrcpad, *mysinkpad;
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static GstElement *dec;
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static GList *events = NULL;
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#define TEST_MSECS_PER_SAMPLE 44100
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#define RESTRICTED_CAPS_RATE 44100
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#define RESTRICTED_CAPS_CHANNELS 6
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static GstStaticPadTemplate sinktemplate_restricted =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, rate=(int)44100, channels=(int)6")
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);
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static GstStaticPadTemplate sinktemplate_with_range =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, rate=(int)[1,44100], channels=(int)[1,6]")
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);
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static GstStaticPadTemplate sinktemplate_default =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, format=(string)S32LE, "
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"rate=(int)[1, 320000], channels=(int)[1, 32],"
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"layout=(string)interleaved")
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);
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static GstStaticPadTemplate srctemplate_default =
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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 ("audio/x-test-custom")
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);
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#define GST_AUDIO_DECODER_TESTER_TYPE gst_audio_decoder_tester_get_type()
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static GType gst_audio_decoder_tester_get_type (void);
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typedef struct _GstAudioDecoderTester GstAudioDecoderTester;
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typedef struct _GstAudioDecoderTesterClass GstAudioDecoderTesterClass;
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struct _GstAudioDecoderTester
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{
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GstAudioDecoder parent;
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gboolean setoutputformat_on_decoding;
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gboolean output_too_many_frames;
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};
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struct _GstAudioDecoderTesterClass
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{
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GstAudioDecoderClass parent_class;
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};
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G_DEFINE_TYPE (GstAudioDecoderTester, gst_audio_decoder_tester,
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GST_TYPE_AUDIO_DECODER);
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static gboolean
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gst_audio_decoder_tester_start (GstAudioDecoder * dec)
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{
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return TRUE;
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}
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static gboolean
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gst_audio_decoder_tester_stop (GstAudioDecoder * dec)
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{
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return TRUE;
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}
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static void
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gst_audio_decoder_tester_flush (GstAudioDecoder * dec, gboolean hard)
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{
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}
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static gboolean
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gst_audio_decoder_tester_set_format (GstAudioDecoder * dec, GstCaps * caps)
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{
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GstAudioDecoderTester *tester = (GstAudioDecoderTester *) dec;
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GstAudioInfo info;
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if (!tester->setoutputformat_on_decoding) {
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caps = gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, "S32LE",
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"channels", G_TYPE_INT, 2, "rate", G_TYPE_INT, 44100,
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"layout", G_TYPE_STRING, "interleaved", NULL);
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gst_audio_info_from_caps (&info, caps);
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gst_caps_unref (caps);
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gst_audio_decoder_set_output_format (dec, &info);
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}
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return TRUE;
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}
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static GstFlowReturn
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gst_audio_decoder_tester_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer)
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{
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GstAudioDecoderTester *tester = (GstAudioDecoderTester *) dec;
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guint8 *data;
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gint size;
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GstMapInfo map;
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GstBuffer *output_buffer;
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if (buffer == NULL)
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return GST_FLOW_OK;
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if (tester->setoutputformat_on_decoding) {
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GstCaps *caps;
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GstAudioInfo info;
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caps = gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, "S32LE",
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"channels", G_TYPE_INT, 2, "rate", G_TYPE_INT, 44100,
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"layout", G_TYPE_STRING, "interleaved", NULL);
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gst_audio_info_from_caps (&info, caps);
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gst_caps_unref (caps);
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gst_audio_decoder_set_output_format (dec, &info);
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}
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gst_buffer_map (buffer, &map, GST_MAP_READ);
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/* the output is SE32LE stereo 44100 Hz */
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size = 2 * 4;
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g_assert (size == sizeof (guint64));
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data = g_malloc0 (size);
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memcpy (data, map.data, sizeof (guint64));
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output_buffer = gst_buffer_new_wrapped (data, size);
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gst_buffer_unmap (buffer, &map);
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if (tester->output_too_many_frames) {
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return gst_audio_decoder_finish_frame (dec, output_buffer, 2);
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} else {
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return gst_audio_decoder_finish_frame (dec, output_buffer, 1);
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}
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}
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static void
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gst_audio_decoder_tester_class_init (GstAudioDecoderTesterClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstAudioDecoderClass *audiosink_class = GST_AUDIO_DECODER_CLASS (klass);
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static GstStaticPadTemplate sink_templ = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK, GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-test-custom"));
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static GstStaticPadTemplate src_templ = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC, GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw"));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_templ));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_templ));
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gst_element_class_set_metadata (element_class,
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"AudioDecoderTester", "Decoder/Audio", "yep", "me");
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audiosink_class->start = gst_audio_decoder_tester_start;
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audiosink_class->stop = gst_audio_decoder_tester_stop;
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audiosink_class->flush = gst_audio_decoder_tester_flush;
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audiosink_class->handle_frame = gst_audio_decoder_tester_handle_frame;
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audiosink_class->set_format = gst_audio_decoder_tester_set_format;
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}
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static void
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gst_audio_decoder_tester_init (GstAudioDecoderTester * tester)
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{
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}
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static gboolean
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_mysinkpad_event (GstPad * pad, GstObject * parent, GstEvent * event)
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{
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events = g_list_append (events, event);
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return TRUE;
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}
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static void
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setup_audiodecodertester (GstStaticPadTemplate * sinktemplate,
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GstStaticPadTemplate * srctemplate)
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{
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if (sinktemplate == NULL)
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sinktemplate = &sinktemplate_default;
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if (srctemplate == NULL)
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srctemplate = &srctemplate_default;
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dec = g_object_new (GST_AUDIO_DECODER_TESTER_TYPE, NULL);
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mysrcpad = gst_check_setup_src_pad (dec, srctemplate);
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mysinkpad = gst_check_setup_sink_pad (dec, sinktemplate);
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gst_pad_set_event_function (mysinkpad, _mysinkpad_event);
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}
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static void
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cleanup_audiodecodertest (void)
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{
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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 (dec);
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gst_check_teardown_sink_pad (dec);
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gst_check_teardown_element (dec);
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}
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static GstBuffer *
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create_test_buffer (guint64 num)
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{
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GstBuffer *buffer;
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guint64 *data = g_malloc (sizeof (guint64));
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*data = num;
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buffer = gst_buffer_new_wrapped (data, sizeof (guint64));
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GST_BUFFER_PTS (buffer) =
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gst_util_uint64_scale_round (num, GST_SECOND, TEST_MSECS_PER_SAMPLE);
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GST_BUFFER_DURATION (buffer) =
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gst_util_uint64_scale_round (1, GST_SECOND, TEST_MSECS_PER_SAMPLE);
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return buffer;
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}
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static void
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send_startup_events (void)
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{
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GstCaps *caps;
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fail_unless (gst_pad_push_event (mysrcpad,
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gst_event_new_stream_start ("randomvalue")));
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/* push caps */
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caps =
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gst_caps_new_simple ("audio/x-test-custom", "channels", G_TYPE_INT, 2,
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"rate", G_TYPE_INT, 44100, NULL);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_caps (caps)));
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gst_caps_unref (caps);
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}
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#define NUM_BUFFERS 1000
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GST_START_TEST (audiodecoder_playback)
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{
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GstSegment segment;
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GstBuffer *buffer;
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guint64 i;
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setup_audiodecodertester (NULL, NULL);
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gst_pad_set_active (mysrcpad, TRUE);
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gst_element_set_state (dec, GST_STATE_PLAYING);
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gst_pad_set_active (mysinkpad, TRUE);
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send_startup_events ();
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/* push a new segment */
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gst_segment_init (&segment, GST_FORMAT_TIME);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
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/* push buffers, the data is actually a number so we can track them */
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for (i = 0; i < NUM_BUFFERS; i++) {
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GstMapInfo map;
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guint64 num;
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buffer = create_test_buffer (i);
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fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
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/* check that buffer was received by our source pad */
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buffer = buffers->data;
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gst_buffer_map (buffer, &map, GST_MAP_READ);
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num = *(guint64 *) map.data;
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fail_unless_equals_uint64 (i, num);
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fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer),
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gst_util_uint64_scale_round (i, GST_SECOND, TEST_MSECS_PER_SAMPLE));
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fail_unless_equals_uint64 (GST_BUFFER_DURATION (buffer),
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gst_util_uint64_scale_round (1, GST_SECOND, TEST_MSECS_PER_SAMPLE));
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gst_buffer_unmap (buffer, &map);
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gst_buffer_unref (buffer);
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buffers = g_list_delete_link (buffers, buffers);
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}
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
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fail_unless (buffers == NULL);
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cleanup_audiodecodertest ();
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}
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GST_END_TEST;
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static void
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check_audiodecoder_negotiation (void)
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{
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gboolean received_caps = FALSE;
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GList *iter;
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for (iter = events; iter; iter = g_list_next (iter)) {
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GstEvent *event = iter->data;
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if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
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GstCaps *caps;
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GstStructure *structure;
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gint channels;
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gint rate;
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gst_event_parse_caps (event, &caps);
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structure = gst_caps_get_structure (caps, 0);
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fail_unless (gst_structure_get_int (structure, "rate", &rate));
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fail_unless (gst_structure_get_int (structure, "channels", &channels));
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fail_unless (rate == 44100, "%d != %d", rate, 44100);
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fail_unless (channels == 2, "%d != %d", channels, 2);
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received_caps = TRUE;
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break;
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}
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}
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fail_unless (received_caps);
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}
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GST_START_TEST (audiodecoder_negotiation_with_buffer)
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{
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GstSegment segment;
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GstBuffer *buffer;
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setup_audiodecodertester (NULL, NULL);
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gst_pad_set_active (mysrcpad, TRUE);
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gst_element_set_state (dec, GST_STATE_PLAYING);
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gst_pad_set_active (mysinkpad, TRUE);
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send_startup_events ();
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/* push a new segment */
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gst_segment_init (&segment, GST_FORMAT_TIME);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
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/* push a buffer event to force audiodecoder to push a caps event */
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buffer = create_test_buffer (0);
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fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
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check_audiodecoder_negotiation ();
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cleanup_audiodecodertest ();
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g_list_free_full (buffers, (GDestroyNotify) gst_buffer_unref);
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}
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GST_END_TEST;
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GST_START_TEST (audiodecoder_negotiation_with_gap_event)
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{
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GstSegment segment;
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setup_audiodecodertester (NULL, NULL);
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gst_pad_set_active (mysrcpad, TRUE);
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gst_element_set_state (dec, GST_STATE_PLAYING);
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gst_pad_set_active (mysinkpad, TRUE);
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send_startup_events ();
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/* push a new segment */
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gst_segment_init (&segment, GST_FORMAT_TIME);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
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/* push a gap event to force audiodecoder to push a caps event */
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_gap (0,
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GST_SECOND)));
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fail_unless (buffers == NULL);
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check_audiodecoder_negotiation ();
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cleanup_audiodecodertest ();
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}
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GST_END_TEST;
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GST_START_TEST (audiodecoder_delayed_negotiation_with_gap_event)
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{
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GstSegment segment;
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setup_audiodecodertester (NULL, NULL);
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((GstAudioDecoderTester *) dec)->setoutputformat_on_decoding = TRUE;
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gst_pad_set_active (mysrcpad, TRUE);
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gst_element_set_state (dec, GST_STATE_PLAYING);
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gst_pad_set_active (mysinkpad, TRUE);
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send_startup_events ();
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/* push a new segment */
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gst_segment_init (&segment, GST_FORMAT_TIME);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
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/* push a gap event to force audiodecoder to push a caps event */
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_gap (0,
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GST_SECOND)));
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fail_unless (buffers == NULL);
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check_audiodecoder_negotiation ();
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cleanup_audiodecodertest ();
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}
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GST_END_TEST;
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static void
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_audiodecoder_flush_events (gboolean send_buffers)
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{
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GstSegment segment;
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GstBuffer *buffer;
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guint64 i;
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GList *events_iter;
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GstMessage *msg;
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setup_audiodecodertester (NULL, NULL);
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gst_pad_set_active (mysrcpad, TRUE);
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gst_element_set_state (dec, GST_STATE_PLAYING);
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gst_pad_set_active (mysinkpad, TRUE);
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send_startup_events ();
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/* push a new segment */
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gst_segment_init (&segment, GST_FORMAT_TIME);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
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if (send_buffers) {
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/* push buffers, the data is actually a number so we can track them */
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for (i = 0; i < NUM_BUFFERS; i++) {
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if (i % 10 == 0) {
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GstTagList *tags;
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tags = gst_tag_list_new (GST_TAG_TRACK_NUMBER, i, NULL);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_tag (tags)));
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} else {
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buffer = create_test_buffer (i);
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fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
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}
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}
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} else {
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/* push sticky event */
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GstTagList *tags;
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tags = gst_tag_list_new (GST_TAG_TRACK_NUMBER, 0, NULL);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_tag (tags)));
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}
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msg =
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gst_message_new_element (GST_OBJECT (mysrcpad),
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gst_structure_new_empty ("test"));
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fail_unless (gst_pad_push_event (mysrcpad,
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gst_event_new_sink_message ("test", msg)));
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gst_message_unref (msg);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
|
|
|
|
events_iter = events;
|
|
/* make sure the usual events have been received */
|
|
{
|
|
GstEvent *sstart = events_iter->data;
|
|
fail_unless (GST_EVENT_TYPE (sstart) == GST_EVENT_STREAM_START);
|
|
events_iter = g_list_next (events_iter);
|
|
}
|
|
if (send_buffers) {
|
|
{
|
|
GstEvent *caps_event = events_iter->data;
|
|
fail_unless (GST_EVENT_TYPE (caps_event) == GST_EVENT_CAPS);
|
|
events_iter = g_list_next (events_iter);
|
|
}
|
|
{
|
|
GstEvent *segment_event = events_iter->data;
|
|
fail_unless (GST_EVENT_TYPE (segment_event) == GST_EVENT_SEGMENT);
|
|
events_iter = g_list_next (events_iter);
|
|
}
|
|
for (int i = 0; i < NUM_BUFFERS / 10; i++) {
|
|
GstEvent *tag_event = events_iter->data;
|
|
fail_unless (GST_EVENT_TYPE (tag_event) == GST_EVENT_TAG);
|
|
events_iter = g_list_next (events_iter);
|
|
}
|
|
}
|
|
{
|
|
GstEvent *eos_event = g_list_last (events_iter)->data;
|
|
|
|
fail_unless (GST_EVENT_TYPE (eos_event) == GST_EVENT_EOS);
|
|
events_iter = g_list_next (events_iter);
|
|
}
|
|
|
|
/* check that EOS was received */
|
|
fail_unless (GST_PAD_IS_EOS (mysrcpad));
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_flush_start ()));
|
|
fail_unless (GST_PAD_IS_EOS (mysrcpad));
|
|
|
|
/* Check that we have tags */
|
|
{
|
|
GstEvent *tags = gst_pad_get_sticky_event (mysrcpad, GST_EVENT_TAG, 0);
|
|
|
|
fail_unless (tags != NULL);
|
|
gst_event_unref (tags);
|
|
}
|
|
|
|
/* Check that we still have a segment set */
|
|
{
|
|
GstEvent *segment =
|
|
gst_pad_get_sticky_event (mysrcpad, GST_EVENT_SEGMENT, 0);
|
|
|
|
fail_unless (segment != NULL);
|
|
gst_event_unref (segment);
|
|
}
|
|
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_flush_stop (TRUE)));
|
|
fail_if (GST_PAD_IS_EOS (mysrcpad));
|
|
|
|
/* Check that the segment was flushed on FLUSH_STOP */
|
|
{
|
|
GstEvent *segment =
|
|
gst_pad_get_sticky_event (mysrcpad, GST_EVENT_SEGMENT, 0);
|
|
|
|
fail_unless (segment == NULL);
|
|
}
|
|
|
|
/* Check the tags were not lost on FLUSH_STOP */
|
|
{
|
|
GstEvent *tags = gst_pad_get_sticky_event (mysrcpad, GST_EVENT_TAG, 0);
|
|
|
|
fail_unless (tags != NULL);
|
|
gst_event_unref (tags);
|
|
|
|
}
|
|
|
|
g_list_free_full (events, (GDestroyNotify) gst_event_unref);
|
|
events = NULL;
|
|
|
|
g_list_free_full (buffers, (GDestroyNotify) gst_buffer_unref);
|
|
buffers = NULL;
|
|
|
|
gst_element_set_state (dec, GST_STATE_NULL);
|
|
cleanup_audiodecodertest ();
|
|
}
|
|
|
|
/* An element should always push its segment before sending EOS */
|
|
GST_START_TEST (audiodecoder_eos_events_no_buffers)
|
|
{
|
|
GstSegment segment;
|
|
setup_audiodecodertester (NULL, NULL);
|
|
|
|
gst_pad_set_active (mysrcpad, TRUE);
|
|
gst_element_set_state (dec, GST_STATE_PLAYING);
|
|
gst_pad_set_active (mysinkpad, TRUE);
|
|
send_startup_events ();
|
|
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
|
|
|
|
fail_unless (GST_PAD_IS_EOS (mysinkpad));
|
|
|
|
{
|
|
GstEvent *segment_event =
|
|
gst_pad_get_sticky_event (mysinkpad, GST_EVENT_SEGMENT, 0);
|
|
fail_unless (segment_event != NULL);
|
|
gst_event_unref (segment_event);
|
|
}
|
|
|
|
gst_element_set_state (dec, GST_STATE_NULL);
|
|
cleanup_audiodecodertest ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (audiodecoder_flush_events_no_buffers)
|
|
{
|
|
_audiodecoder_flush_events (FALSE);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (audiodecoder_flush_events)
|
|
{
|
|
_audiodecoder_flush_events (TRUE);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
|
|
GST_START_TEST (audiodecoder_buffer_after_segment)
|
|
{
|
|
GstSegment segment;
|
|
GstBuffer *buffer;
|
|
guint64 i;
|
|
GstClockTime pos;
|
|
|
|
setup_audiodecodertester (NULL, NULL);
|
|
|
|
gst_pad_set_active (mysrcpad, TRUE);
|
|
gst_element_set_state (dec, GST_STATE_PLAYING);
|
|
gst_pad_set_active (mysinkpad, TRUE);
|
|
|
|
send_startup_events ();
|
|
|
|
/* push a new segment */
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
segment.stop = GST_SECOND;
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
|
|
|
|
/* push buffers, the data is actually a number so we can track them */
|
|
i = 0;
|
|
pos = 0;
|
|
while (pos < GST_SECOND) {
|
|
GstMapInfo map;
|
|
guint64 num;
|
|
|
|
buffer = create_test_buffer (i);
|
|
pos = GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer);
|
|
|
|
fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
|
|
|
|
/* check that buffer was received by our source pad */
|
|
buffer = buffers->data;
|
|
|
|
gst_buffer_map (buffer, &map, GST_MAP_READ);
|
|
|
|
num = *(guint64 *) map.data;
|
|
fail_unless_equals_uint64 (i, num);
|
|
fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer),
|
|
gst_util_uint64_scale_round (i, GST_SECOND, TEST_MSECS_PER_SAMPLE));
|
|
fail_unless_equals_uint64 (GST_BUFFER_DURATION (buffer),
|
|
gst_util_uint64_scale_round (1, GST_SECOND, TEST_MSECS_PER_SAMPLE));
|
|
|
|
gst_buffer_unmap (buffer, &map);
|
|
|
|
gst_buffer_unref (buffer);
|
|
buffers = g_list_delete_link (buffers, buffers);
|
|
i++;
|
|
}
|
|
|
|
/* this buffer is after the segment */
|
|
buffer = create_test_buffer (i++);
|
|
fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_EOS);
|
|
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
|
|
|
|
fail_unless (buffers == NULL);
|
|
|
|
cleanup_audiodecodertest ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (audiodecoder_output_too_many_frames)
|
|
{
|
|
GstSegment segment;
|
|
GstBuffer *buffer;
|
|
guint64 i;
|
|
|
|
setup_audiodecodertester (NULL, NULL);
|
|
|
|
((GstAudioDecoderTester *) dec)->output_too_many_frames = TRUE;
|
|
|
|
gst_pad_set_active (mysrcpad, TRUE);
|
|
gst_element_set_state (dec, GST_STATE_PLAYING);
|
|
gst_pad_set_active (mysinkpad, TRUE);
|
|
|
|
send_startup_events ();
|
|
|
|
/* push a new segment */
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
|
|
|
|
/* push buffers, the data is actually a number so we can track them */
|
|
for (i = 0; i < 3; i++) {
|
|
GstMapInfo map;
|
|
guint64 num;
|
|
|
|
buffer = create_test_buffer (i);
|
|
|
|
fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
|
|
|
|
/* check that buffer was received by our source pad */
|
|
buffer = buffers->data;
|
|
|
|
gst_buffer_map (buffer, &map, GST_MAP_READ);
|
|
|
|
num = *(guint64 *) map.data;
|
|
fail_unless_equals_uint64 (i, num);
|
|
fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer),
|
|
gst_util_uint64_scale_round (i, GST_SECOND, TEST_MSECS_PER_SAMPLE));
|
|
fail_unless_equals_uint64 (GST_BUFFER_DURATION (buffer),
|
|
gst_util_uint64_scale_round (1, GST_SECOND, TEST_MSECS_PER_SAMPLE));
|
|
|
|
gst_buffer_unmap (buffer, &map);
|
|
|
|
gst_buffer_unref (buffer);
|
|
buffers = g_list_delete_link (buffers, buffers);
|
|
}
|
|
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
|
|
|
|
fail_unless (buffers == NULL);
|
|
|
|
cleanup_audiodecodertest ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (audiodecoder_query_caps_with_fixed_caps_peer)
|
|
{
|
|
GstCaps *caps;
|
|
GstCaps *filter;
|
|
GstStructure *structure;
|
|
gint rate, channels;
|
|
|
|
setup_audiodecodertester (&sinktemplate_restricted, NULL);
|
|
|
|
gst_pad_set_active (mysrcpad, TRUE);
|
|
gst_element_set_state (dec, GST_STATE_PLAYING);
|
|
gst_pad_set_active (mysinkpad, TRUE);
|
|
|
|
caps = gst_pad_peer_query_caps (mysrcpad, NULL);
|
|
fail_unless (caps != NULL);
|
|
|
|
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));
|
|
|
|
/* match our restricted caps values */
|
|
fail_unless (channels == RESTRICTED_CAPS_CHANNELS);
|
|
fail_unless (rate == RESTRICTED_CAPS_RATE);
|
|
gst_caps_unref (caps);
|
|
|
|
filter = gst_caps_new_simple ("audio/x-custom-test", "rate", G_TYPE_INT,
|
|
10000, "channels", G_TYPE_INT, 12, NULL);
|
|
caps = gst_pad_peer_query_caps (mysrcpad, filter);
|
|
fail_unless (caps != NULL);
|
|
fail_unless (gst_caps_is_empty (caps));
|
|
gst_caps_unref (caps);
|
|
gst_caps_unref (filter);
|
|
|
|
cleanup_audiodecodertest ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static void
|
|
_get_int_range (GstStructure * s, const gchar * field, gint * min_v,
|
|
gint * max_v)
|
|
{
|
|
const GValue *value;
|
|
|
|
value = gst_structure_get_value (s, field);
|
|
fail_unless (value != NULL);
|
|
fail_unless (GST_VALUE_HOLDS_INT_RANGE (value));
|
|
|
|
*min_v = gst_value_get_int_range_min (value);
|
|
*max_v = gst_value_get_int_range_max (value);
|
|
}
|
|
|
|
GST_START_TEST (audiodecoder_query_caps_with_range_caps_peer)
|
|
{
|
|
GstCaps *caps;
|
|
GstCaps *filter;
|
|
GstStructure *structure;
|
|
gint rate, channels;
|
|
gint rate_min, channels_min;
|
|
gint rate_max, channels_max;
|
|
|
|
setup_audiodecodertester (&sinktemplate_with_range, NULL);
|
|
|
|
gst_pad_set_active (mysrcpad, TRUE);
|
|
gst_element_set_state (dec, GST_STATE_PLAYING);
|
|
gst_pad_set_active (mysinkpad, TRUE);
|
|
|
|
caps = gst_pad_peer_query_caps (mysrcpad, NULL);
|
|
fail_unless (caps != NULL);
|
|
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
_get_int_range (structure, "rate", &rate_min, &rate_max);
|
|
_get_int_range (structure, "channels", &channels_min, &channels_max);
|
|
fail_unless (rate_min == 1);
|
|
fail_unless (rate_max == RESTRICTED_CAPS_RATE);
|
|
fail_unless (channels_min == 1);
|
|
fail_unless (channels_max == RESTRICTED_CAPS_CHANNELS);
|
|
gst_caps_unref (caps);
|
|
|
|
/* query with a fixed filter */
|
|
filter = gst_caps_new_simple ("audio/x-test-custom", "rate", G_TYPE_INT,
|
|
RESTRICTED_CAPS_RATE, "channels", G_TYPE_INT, RESTRICTED_CAPS_CHANNELS,
|
|
NULL);
|
|
caps = gst_pad_peer_query_caps (mysrcpad, filter);
|
|
fail_unless (caps != NULL);
|
|
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));
|
|
fail_unless (rate == RESTRICTED_CAPS_RATE);
|
|
fail_unless (channels == RESTRICTED_CAPS_CHANNELS);
|
|
gst_caps_unref (caps);
|
|
gst_caps_unref (filter);
|
|
|
|
/* query with a fixed filter that will lead to empty result */
|
|
filter = gst_caps_new_simple ("audio/x-test-custom", "rate", G_TYPE_INT,
|
|
10000, "channels", G_TYPE_INT, 12, NULL);
|
|
caps = gst_pad_peer_query_caps (mysrcpad, filter);
|
|
fail_unless (caps != NULL);
|
|
fail_unless (gst_caps_is_empty (caps));
|
|
gst_caps_unref (caps);
|
|
gst_caps_unref (filter);
|
|
|
|
cleanup_audiodecodertest ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
#define GETCAPS_CAPS_STR "audio/x-test-custom, somefield=(string)getcaps"
|
|
static GstCaps *
|
|
_custom_audio_decoder_getcaps (GstAudioDecoder * dec, GstCaps * filter)
|
|
{
|
|
return gst_caps_from_string (GETCAPS_CAPS_STR);
|
|
}
|
|
|
|
GST_START_TEST (audiodecoder_query_caps_with_custom_getcaps)
|
|
{
|
|
GstCaps *caps;
|
|
GstAudioDecoderClass *klass;
|
|
GstCaps *expected_caps;
|
|
|
|
setup_audiodecodertester (&sinktemplate_restricted, NULL);
|
|
|
|
klass = GST_AUDIO_DECODER_CLASS (GST_AUDIO_DECODER_GET_CLASS (dec));
|
|
klass->getcaps = _custom_audio_decoder_getcaps;
|
|
|
|
gst_pad_set_active (mysrcpad, TRUE);
|
|
gst_element_set_state (dec, GST_STATE_PLAYING);
|
|
gst_pad_set_active (mysinkpad, TRUE);
|
|
|
|
caps = gst_pad_peer_query_caps (mysrcpad, NULL);
|
|
fail_unless (caps != NULL);
|
|
|
|
expected_caps = gst_caps_from_string (GETCAPS_CAPS_STR);
|
|
fail_unless (gst_caps_is_equal (expected_caps, caps));
|
|
gst_caps_unref (expected_caps);
|
|
gst_caps_unref (caps);
|
|
|
|
cleanup_audiodecodertest ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
|
|
static Suite *
|
|
gst_audiodecoder_suite (void)
|
|
{
|
|
Suite *s = suite_create ("GstAudioDecoder");
|
|
TCase *tc = tcase_create ("general");
|
|
|
|
suite_add_tcase (s, tc);
|
|
tcase_add_test (tc, audiodecoder_playback);
|
|
tcase_add_test (tc, audiodecoder_flush_events_no_buffers);
|
|
tcase_add_test (tc, audiodecoder_eos_events_no_buffers);
|
|
tcase_add_test (tc, audiodecoder_flush_events);
|
|
tcase_add_test (tc, audiodecoder_negotiation_with_buffer);
|
|
tcase_add_test (tc, audiodecoder_negotiation_with_gap_event);
|
|
tcase_add_test (tc, audiodecoder_delayed_negotiation_with_gap_event);
|
|
tcase_add_test (tc, audiodecoder_buffer_after_segment);
|
|
tcase_add_test (tc, audiodecoder_output_too_many_frames);
|
|
|
|
tcase_add_test (tc, audiodecoder_query_caps_with_fixed_caps_peer);
|
|
tcase_add_test (tc, audiodecoder_query_caps_with_range_caps_peer);
|
|
tcase_add_test (tc, audiodecoder_query_caps_with_custom_getcaps);
|
|
|
|
return s;
|
|
}
|
|
|
|
GST_CHECK_MAIN (gst_audiodecoder);
|