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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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338 lines
10 KiB
C
338 lines
10 KiB
C
/* GStreamer
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*
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* Copyright (C) 2009 Nokia Corporation and its subsidary(-ies)
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* contact: <stefan.kost@nokia.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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <gst/check/gstcheck.h>
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/* For ease of programming we use globals to keep refs for our floating
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* src and sink pads we create; otherwise we always have to do get_pad,
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* get_peer, and then remove references in every test function */
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static GstPad *mysrcpad, *mysinkpad;
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/* we also have a list of src buffers */
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static GList *inbuffers = NULL;
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static gint num_dropped = 0;
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#define RTP_CAPS_STRING \
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"application/x-rtp, " \
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"media = (string)audio, " \
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"payload = (int) 0, " \
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"clock-rate = (int) 8000, " \
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"encoding-name = (string)PCMU"
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#define RTP_FRAME_SIZE 20
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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 ("application/x-rtp")
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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 ("application/x-rtp, "
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"clock-rate = (int) [ 1, 2147483647 ]")
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);
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static void
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buffer_dropped (gpointer data, GstMiniObject * obj)
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{
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GST_DEBUG ("dropping buffer %p", obj);
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num_dropped++;
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}
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static GstElement *
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setup_jitterbuffer (gint num_buffers)
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{
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GstElement *jitterbuffer;
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GstClock *clock;
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GstBuffer *buffer;
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GstCaps *caps;
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/* a 20 sample audio block (2,5 ms) generated with
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* gst-launch audiotestsrc wave=silence blocksize=40 num-buffers=3 !
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* "audio/x-raw,channels=1,rate=8000" ! mulawenc ! rtppcmupay !
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* fakesink dump=1
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*/
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guint8 in[] = { /* first 4 bytes are rtp-header, next 4 bytes are timestamp */
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0x80, 0x80, 0x1c, 0x24, 0x46, 0xcd, 0xb7, 0x11, 0x3c, 0x3a, 0x7c, 0x5b,
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0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
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0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff
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};
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GstClockTime ts = G_GUINT64_CONSTANT (0);
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GstClockTime tso = gst_util_uint64_scale (RTP_FRAME_SIZE, GST_SECOND, 8000);
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/*guint latency = GST_TIME_AS_MSECONDS (num_buffers * tso); */
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gint i;
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GST_DEBUG ("setup_jitterbuffer");
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jitterbuffer = gst_check_setup_element ("gstrtpjitterbuffer");
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/* we need a clock here */
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clock = gst_system_clock_obtain ();
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gst_element_set_clock (jitterbuffer, clock);
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gst_object_unref (clock);
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/* setup latency */
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/* latency would be 7 for 3 buffers here, default is 200
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g_object_set (G_OBJECT (jitterbuffer), "latency", latency, NULL);
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GST_INFO_OBJECT (jitterbuffer, "set latency to %u ms", latency);
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*/
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mysrcpad = gst_check_setup_src_pad (jitterbuffer, &srctemplate);
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mysinkpad = gst_check_setup_sink_pad (jitterbuffer, &sinktemplate);
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gst_pad_set_active (mysrcpad, TRUE);
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gst_pad_set_active (mysinkpad, TRUE);
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/* create n buffers */
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caps = gst_caps_from_string (RTP_CAPS_STRING);
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for (i = 0; i < num_buffers; i++) {
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buffer = gst_buffer_new_and_alloc (sizeof (in));
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gst_buffer_fill (buffer, 0, in, sizeof (in));
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GST_BUFFER_TIMESTAMP (buffer) = ts;
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GST_BUFFER_DURATION (buffer) = tso;
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gst_mini_object_weak_ref (GST_MINI_OBJECT (buffer), buffer_dropped, NULL);
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GST_DEBUG ("created buffer: %p", buffer);
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if (!i)
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GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
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inbuffers = g_list_append (inbuffers, buffer);
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/* hackish way to update the rtp header */
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in[1] = 0x00;
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in[3]++; /* seqnumber */
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in[7] += RTP_FRAME_SIZE; /* inc. timestamp with framesize */
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ts += tso;
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}
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gst_caps_unref (caps);
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num_dropped = 0;
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return jitterbuffer;
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}
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static GstStateChangeReturn
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start_jitterbuffer (GstElement * jitterbuffer)
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{
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GstStateChangeReturn ret;
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GstClockTime now;
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GstClock *clock;
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clock = gst_element_get_clock (jitterbuffer);
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now = gst_clock_get_time (clock);
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gst_object_unref (clock);
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gst_element_set_base_time (jitterbuffer, now);
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ret = gst_element_set_state (jitterbuffer, GST_STATE_PLAYING);
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return ret;
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}
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static void
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cleanup_jitterbuffer (GstElement * jitterbuffer)
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{
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GST_DEBUG ("cleanup_jitterbuffer");
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g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
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g_list_free (buffers);
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buffers = NULL;
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g_list_free (inbuffers);
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inbuffers = NULL;
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gst_pad_set_active (mysrcpad, FALSE);
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gst_pad_set_active (mysinkpad, FALSE);
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gst_check_teardown_src_pad (jitterbuffer);
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gst_check_teardown_sink_pad (jitterbuffer);
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gst_check_teardown_element (jitterbuffer);
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}
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static void
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check_jitterbuffer_results (GstElement * jitterbuffer, gint num_buffers)
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{
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GstBuffer *buffer;
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GList *node;
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GstClockTime ts = G_GUINT64_CONSTANT (0);
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GstClockTime tso = gst_util_uint64_scale (RTP_FRAME_SIZE, GST_SECOND, 8000);
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guint8 *data;
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guint16 prev_sn = 0, cur_sn;
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guint32 prev_ts = 0, cur_ts;
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/* sleep for twice the latency */
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g_usleep (400 * 1000);
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GST_INFO ("of %d buffer %d/%d received/dropped", num_buffers,
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g_list_length (buffers), num_dropped);
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/* if this fails, not all buffers have been processed */
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fail_unless_equals_int ((g_list_length (buffers) + num_dropped), num_buffers);
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/* check the buffer list */
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fail_unless_equals_int (g_list_length (buffers), num_buffers);
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for (node = buffers; node; node = g_list_next (node)) {
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fail_if ((buffer = (GstBuffer *) node->data) == NULL);
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fail_if (GST_BUFFER_TIMESTAMP (buffer) != ts);
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data = gst_buffer_map (buffer, NULL, NULL, GST_MAP_READ);
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cur_sn = ((guint16) data[2] << 8) | data[3];
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cur_ts = ((guint32) data[4] << 24) | ((guint32) data[5] << 16) |
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((guint32) data[6] << 8) | data[7];
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gst_buffer_unmap (buffer, data, -1);
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if (node != buffers) {
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fail_unless (cur_sn > prev_sn);
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fail_unless (cur_ts > prev_ts);
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prev_sn = cur_sn;
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prev_ts = cur_ts;
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}
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ts += tso;
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}
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}
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GST_START_TEST (test_push_forward_seq)
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{
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GstElement *jitterbuffer;
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const guint num_buffers = 3;
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GstBuffer *buffer;
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GList *node;
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jitterbuffer = setup_jitterbuffer (num_buffers);
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fail_unless (start_jitterbuffer (jitterbuffer)
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== GST_STATE_CHANGE_SUCCESS, "could not set to playing");
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/* push buffers: 0,1,2, */
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for (node = inbuffers; node; node = g_list_next (node)) {
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buffer = (GstBuffer *) node->data;
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fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
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}
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/* check the buffer list */
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check_jitterbuffer_results (jitterbuffer, num_buffers);
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/* cleanup */
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cleanup_jitterbuffer (jitterbuffer);
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}
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GST_END_TEST;
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GST_START_TEST (test_push_backward_seq)
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{
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GstElement *jitterbuffer;
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const guint num_buffers = 4;
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GstBuffer *buffer;
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GList *node;
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jitterbuffer = setup_jitterbuffer (num_buffers);
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fail_unless (start_jitterbuffer (jitterbuffer)
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== GST_STATE_CHANGE_SUCCESS, "could not set to playing");
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/* push buffers: 0,3,2,1 */
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buffer = (GstBuffer *) inbuffers->data;
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fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
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for (node = g_list_last (inbuffers); node != inbuffers;
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node = g_list_previous (node)) {
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buffer = (GstBuffer *) node->data;
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fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
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}
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/* check the buffer list */
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check_jitterbuffer_results (jitterbuffer, num_buffers);
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/* cleanup */
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cleanup_jitterbuffer (jitterbuffer);
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}
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GST_END_TEST;
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GST_START_TEST (test_push_unordered)
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{
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GstElement *jitterbuffer;
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const guint num_buffers = 4;
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GstBuffer *buffer;
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jitterbuffer = setup_jitterbuffer (num_buffers);
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fail_unless (start_jitterbuffer (jitterbuffer)
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== GST_STATE_CHANGE_SUCCESS, "could not set to playing");
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/* push buffers; 0,2,1,3 */
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buffer = (GstBuffer *) inbuffers->data;
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fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
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buffer = g_list_nth_data (inbuffers, 2);
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fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
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buffer = g_list_nth_data (inbuffers, 1);
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fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
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buffer = g_list_nth_data (inbuffers, 3);
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fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
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/* check the buffer list */
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check_jitterbuffer_results (jitterbuffer, num_buffers);
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/* cleanup */
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cleanup_jitterbuffer (jitterbuffer);
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}
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GST_END_TEST;
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GST_START_TEST (test_basetime)
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{
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GstElement *jitterbuffer;
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const guint num_buffers = 3;
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GstBuffer *buffer;
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GList *node;
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GstClockTime tso = gst_util_uint64_scale (RTP_FRAME_SIZE, GST_SECOND, 8000);
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jitterbuffer = setup_jitterbuffer (num_buffers);
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fail_unless (start_jitterbuffer (jitterbuffer)
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== GST_STATE_CHANGE_SUCCESS, "could not set to playing");
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/* push buffers: 2,1,0 */
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for (node = g_list_last (inbuffers); node; node = g_list_previous (node)) {
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buffer = (GstBuffer *) node->data;
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fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
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}
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/* sleep for twice the latency */
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g_usleep (400 * 1000);
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/* if this fails, not all buffers have been processed */
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fail_unless_equals_int ((g_list_length (buffers) + num_dropped), num_buffers);
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buffer = (GstBuffer *) buffers->data;
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fail_unless (GST_BUFFER_TIMESTAMP (buffer) != (num_buffers * tso));
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/* cleanup */
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cleanup_jitterbuffer (jitterbuffer);
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}
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GST_END_TEST;
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static Suite *
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rtpjitterbuffer_suite (void)
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{
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Suite *s = suite_create ("rtpjitterbuffer");
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TCase *tc_chain = tcase_create ("general");
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suite_add_tcase (s, tc_chain);
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tcase_add_test (tc_chain, test_push_forward_seq);
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tcase_add_test (tc_chain, test_push_backward_seq);
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tcase_add_test (tc_chain, test_push_unordered);
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tcase_add_test (tc_chain, test_basetime);
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/* FIXME: test buffer lists */
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return s;
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}
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GST_CHECK_MAIN (rtpjitterbuffer);
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