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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1176 lines
40 KiB
C
1176 lines
40 KiB
C
/* GStreamer unit tests for flvmux
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*
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* Copyright (C) 2009 Tim-Philipp Müller <tim centricular net>
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* Copyright (C) 2016 Havard Graff <havard@pexip.com>
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* Copyright (C) 2016 David Buchmann <david@pexip.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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#ifdef HAVE_VALGRIND
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# include <valgrind/valgrind.h>
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#endif
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#include <gst/check/gstcheck.h>
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#include <gst/check/gstharness.h>
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#include <gst/gst.h>
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static GstBusSyncReply
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error_cb (GstBus * bus, GstMessage * msg, gpointer user_data)
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{
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if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR) {
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GError *err = NULL;
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gchar *dbg = NULL;
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gst_message_parse_error (msg, &err, &dbg);
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g_error ("ERROR: %s\n%s\n", err->message, dbg);
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}
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return GST_BUS_PASS;
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}
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static void
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handoff_cb (GstElement * element, GstBuffer * buf, GstPad * pad,
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gint * p_counter)
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{
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*p_counter += 1;
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GST_LOG ("counter = %d", *p_counter);
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}
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static void
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mux_pcm_audio (guint num_buffers, guint repeat)
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{
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GstElement *src, *sink, *flvmux, *conv, *pipeline;
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GstPad *sinkpad, *srcpad;
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gint counter;
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GST_LOG ("num_buffers = %u", num_buffers);
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pipeline = gst_pipeline_new ("pipeline");
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fail_unless (pipeline != NULL, "Failed to create pipeline!");
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/* kids, don't use a sync handler for this at home, really; we do because
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* we just want to abort and nothing else */
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gst_bus_set_sync_handler (GST_ELEMENT_BUS (pipeline), error_cb, NULL, NULL);
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src = gst_element_factory_make ("audiotestsrc", "audiotestsrc");
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fail_unless (src != NULL, "Failed to create 'audiotestsrc' element!");
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g_object_set (src, "num-buffers", num_buffers, NULL);
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conv = gst_element_factory_make ("audioconvert", "audioconvert");
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fail_unless (conv != NULL, "Failed to create 'audioconvert' element!");
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flvmux = gst_element_factory_make ("flvmux", "flvmux");
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fail_unless (flvmux != NULL, "Failed to create 'flvmux' element!");
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sink = gst_element_factory_make ("fakesink", "fakesink");
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fail_unless (sink != NULL, "Failed to create 'fakesink' element!");
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g_object_set (sink, "signal-handoffs", TRUE, NULL);
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g_signal_connect (sink, "handoff", G_CALLBACK (handoff_cb), &counter);
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gst_bin_add_many (GST_BIN (pipeline), src, conv, flvmux, sink, NULL);
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fail_unless (gst_element_link (src, conv));
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fail_unless (gst_element_link (flvmux, sink));
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/* now link the elements */
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sinkpad = gst_element_request_pad_simple (flvmux, "audio");
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fail_unless (sinkpad != NULL, "Could not get audio request pad");
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srcpad = gst_element_get_static_pad (conv, "src");
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fail_unless (srcpad != NULL, "Could not get audioconvert's source pad");
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fail_unless_equals_int (gst_pad_link (srcpad, sinkpad), GST_PAD_LINK_OK);
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gst_object_unref (srcpad);
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gst_object_unref (sinkpad);
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do {
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GstStateChangeReturn state_ret;
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GstMessage *msg;
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GST_LOG ("repeat=%d", repeat);
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counter = 0;
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state_ret = gst_element_set_state (pipeline, GST_STATE_PAUSED);
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fail_unless (state_ret != GST_STATE_CHANGE_FAILURE);
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if (state_ret == GST_STATE_CHANGE_ASYNC) {
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GST_LOG ("waiting for pipeline to reach PAUSED state");
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state_ret = gst_element_get_state (pipeline, NULL, NULL, -1);
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fail_unless_equals_int (state_ret, GST_STATE_CHANGE_SUCCESS);
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}
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GST_LOG ("PAUSED, let's do the rest of it");
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state_ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
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fail_unless (state_ret != GST_STATE_CHANGE_FAILURE);
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msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
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fail_unless (msg != NULL, "Expected EOS message on bus!");
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GST_LOG ("EOS");
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gst_message_unref (msg);
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/* should have some output */
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fail_unless (counter > 2);
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fail_unless_equals_int (gst_element_set_state (pipeline, GST_STATE_NULL),
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GST_STATE_CHANGE_SUCCESS);
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/* repeat = test re-usability */
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--repeat;
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} while (repeat > 0);
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gst_object_unref (pipeline);
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}
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GST_START_TEST (test_index_writing)
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{
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/* note: there's a magic 128 value in flvmux when doing index writing */
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mux_pcm_audio (__i__ * 33 + 1, 2);
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}
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GST_END_TEST;
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static GstBuffer *
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create_buffer (guint8 * data, gsize size,
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GstClockTime timestamp, GstClockTime duration)
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{
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GstBuffer *buf = gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY,
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data, size, 0, size, NULL, NULL);
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GST_BUFFER_PTS (buf) = timestamp;
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GST_BUFFER_DTS (buf) = timestamp;
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GST_BUFFER_DURATION (buf) = duration;
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GST_BUFFER_OFFSET (buf) = 0;
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GST_BUFFER_OFFSET_END (buf) = 0;
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return buf;
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}
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guint8 speex_hdr0[] = {
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0x53, 0x70, 0x65, 0x65, 0x78, 0x20, 0x20, 0x20,
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0x31, 0x2e, 0x32, 0x72, 0x63, 0x31, 0x00, 0x00,
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0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
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0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00,
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0x50, 0x00, 0x00, 0x00, 0x80, 0x3e, 0x00, 0x00,
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0x01, 0x00, 0x00, 0x00, 0x04, 0x00, 0x00, 0x00,
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0x01, 0x00, 0x00, 0x00, 0xff, 0xff, 0xff, 0xff,
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0x40, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
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0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
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0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
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};
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guint8 speex_hdr1[] = {
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0x1f, 0x00, 0x00, 0x00, 0x45, 0x6e, 0x63, 0x6f,
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0x64, 0x65, 0x64, 0x20, 0x77, 0x69, 0x74, 0x68,
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0x20, 0x47, 0x53, 0x74, 0x72, 0x65, 0x61, 0x6d,
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0x65, 0x72, 0x20, 0x53, 0x70, 0x65, 0x65, 0x78,
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0x65, 0x6e, 0x63, 0x00, 0x00, 0x00, 0x00, 0x01
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};
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guint8 speex_buf[] = {
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0x36, 0x9d, 0x1b, 0x9a, 0x20, 0x00, 0x01, 0x68,
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0xe8, 0xe8, 0xe8, 0xe8, 0xe8, 0xe8, 0xe8, 0x84,
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0x00, 0xb4, 0x74, 0x74, 0x74, 0x74, 0x74, 0x74,
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0x74, 0x42, 0x00, 0x5a, 0x3a, 0x3a, 0x3a, 0x3a,
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0x3a, 0x3a, 0x3a, 0x21, 0x00, 0x2d, 0x1d, 0x1d,
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0x1d, 0x1d, 0x1d, 0x1d, 0x1d, 0x1b, 0x3b, 0x60,
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0xab, 0xab, 0xab, 0xab, 0xab, 0x0a, 0xba, 0xba,
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0xba, 0xba, 0xb0, 0xab, 0xab, 0xab, 0xab, 0xab,
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0x0a, 0xba, 0xba, 0xba, 0xba, 0xb7
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};
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guint8 h264_buf[] = {
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0x00, 0x00, 0x00, 0x0b, 0x67, 0x42, 0xc0, 0x0c,
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0x95, 0xa7, 0x20, 0x1e, 0x11, 0x08, 0xd4, 0x00,
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0x00, 0x00, 0x04, 0x68, 0xce, 0x3c, 0x80, 0x00,
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0x00, 0x00, 0x55, 0x65, 0xb8, 0x04, 0x0e, 0x7e,
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0x1f, 0x22, 0x60, 0x34, 0x01, 0xe2, 0x00, 0x3c,
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0xe1, 0xfc, 0x91, 0x40, 0xa6, 0x9e, 0x07, 0x42,
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0x56, 0x44, 0x73, 0x75, 0x40, 0x9f, 0x0c, 0x87,
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0x83, 0xc9, 0x52, 0x60, 0x6d, 0xd8, 0x98, 0x01,
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0x16, 0xbd, 0x0f, 0xa6, 0xaf, 0x75, 0x83, 0xdd,
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0xfa, 0xe7, 0x8f, 0xe3, 0x58, 0x10, 0x0f, 0x5c,
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0x18, 0x2f, 0x41, 0x40, 0x23, 0x0b, 0x03, 0x70,
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0x00, 0xff, 0xe4, 0xa6, 0x7d, 0x7f, 0x3f, 0x76,
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0x01, 0xd0, 0x98, 0x2a, 0x0c, 0xb8, 0x02, 0x32,
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0xbc, 0x56, 0xfd, 0x34, 0x4f, 0xcf, 0xfe, 0xa0,
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};
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GST_START_TEST (test_speex_streamable)
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{
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GstBuffer *buf;
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GstMapInfo map = GST_MAP_INFO_INIT;
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GstCaps *caps = gst_caps_new_simple ("audio/x-speex",
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"rate", G_TYPE_INT, 16000,
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"channels", G_TYPE_INT, 1,
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NULL);
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const GstClockTime base_time = 123456789;
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const GstClockTime duration_ms = 20;
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const GstClockTime duration = duration_ms * GST_MSECOND;
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GstHarness *h = gst_harness_new_with_padnames ("flvmux", "audio", "src");
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gst_harness_set_src_caps (h, caps);
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g_object_set (h->element, "streamable", 1, NULL);
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/* push speex header0 */
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gst_harness_push (h, create_buffer (speex_hdr0,
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sizeof (speex_hdr0), base_time, 0));
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/* push speex header1 */
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gst_harness_push (h, create_buffer (speex_hdr1,
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sizeof (speex_hdr1), base_time, 0));
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/* push speex data */
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gst_harness_push (h, create_buffer (speex_buf,
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sizeof (speex_buf), base_time, duration));
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/* push speex data 2 */
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gst_harness_push (h, create_buffer (speex_buf,
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sizeof (speex_buf), base_time + duration, duration));
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/* pull out stream-start event */
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gst_event_unref (gst_harness_pull_event (h));
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/* pull out caps event */
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gst_event_unref (gst_harness_pull_event (h));
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/* pull out segment event and verify we are using GST_FORMAT_TIME */
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{
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GstEvent *event = gst_harness_pull_event (h);
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const GstSegment *segment;
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gst_event_parse_segment (event, &segment);
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fail_unless_equals_int (GST_FORMAT_TIME, segment->format);
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gst_event_unref (event);
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}
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/* pull FLV header buffer */
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buf = gst_harness_pull (h);
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gst_buffer_unref (buf);
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/* pull Metadata buffer */
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buf = gst_harness_pull (h);
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gst_buffer_unref (buf);
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/* pull header0 */
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buf = gst_harness_pull (h);
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fail_unless_equals_uint64 (base_time, GST_BUFFER_PTS (buf));
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fail_unless_equals_uint64 (GST_CLOCK_TIME_NONE, GST_BUFFER_DTS (buf));
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fail_unless_equals_uint64 (0, GST_BUFFER_DURATION (buf));
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gst_buffer_map (buf, &map, GST_MAP_READ);
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/* 0x08 means it is audio */
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fail_unless_equals_int (0x08, map.data[0]);
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/* timestamp should be starting from 0 */
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fail_unless_equals_int (0x00, map.data[6]);
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/* 0xb2 means Speex, 16000Hz, Mono */
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fail_unless_equals_int (0xb2, map.data[11]);
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/* verify content is intact */
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fail_unless_equals_int (0, memcmp (&map.data[12], speex_hdr0,
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sizeof (speex_hdr0)));
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gst_buffer_unmap (buf, &map);
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gst_buffer_unref (buf);
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/* pull header1 */
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buf = gst_harness_pull (h);
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fail_unless_equals_uint64 (base_time, GST_BUFFER_PTS (buf));
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fail_unless_equals_uint64 (GST_CLOCK_TIME_NONE, GST_BUFFER_DTS (buf));
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fail_unless_equals_uint64 (0, GST_BUFFER_DURATION (buf));
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gst_buffer_map (buf, &map, GST_MAP_READ);
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/* 0x08 means it is audio */
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fail_unless_equals_int (0x08, map.data[0]);
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/* timestamp should be starting from 0 */
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fail_unless_equals_int (0x00, map.data[6]);
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/* 0xb2 means Speex, 16000Hz, Mono */
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fail_unless_equals_int (0xb2, map.data[11]);
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/* verify content is intact */
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fail_unless_equals_int (0, memcmp (&map.data[12], speex_hdr1,
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sizeof (speex_hdr1)));
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gst_buffer_unmap (buf, &map);
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gst_buffer_unref (buf);
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/* pull data */
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buf = gst_harness_pull (h);
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fail_unless_equals_uint64 (base_time, GST_BUFFER_PTS (buf));
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fail_unless_equals_uint64 (GST_CLOCK_TIME_NONE, GST_BUFFER_DTS (buf));
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fail_unless_equals_uint64 (duration, GST_BUFFER_DURATION (buf));
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fail_unless_equals_uint64 (GST_BUFFER_OFFSET_NONE, GST_BUFFER_OFFSET (buf));
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fail_unless_equals_uint64 (GST_BUFFER_OFFSET_NONE,
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GST_BUFFER_OFFSET_END (buf));
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gst_buffer_map (buf, &map, GST_MAP_READ);
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/* 0x08 means it is audio */
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fail_unless_equals_int (0x08, map.data[0]);
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/* timestamp should be starting from 0 */
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fail_unless_equals_int (0x00, map.data[6]);
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/* 0xb2 means Speex, 16000Hz, Mono */
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fail_unless_equals_int (0xb2, map.data[11]);
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/* verify content is intact */
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fail_unless_equals_int (0, memcmp (&map.data[12], speex_buf,
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sizeof (speex_buf)));
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gst_buffer_unmap (buf, &map);
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gst_buffer_unref (buf);
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/* pull data */
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buf = gst_harness_pull (h);
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fail_unless_equals_uint64 (base_time + duration, GST_BUFFER_PTS (buf));
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fail_unless_equals_uint64 (GST_CLOCK_TIME_NONE, GST_BUFFER_DTS (buf));
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fail_unless_equals_uint64 (duration, GST_BUFFER_DURATION (buf));
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fail_unless_equals_uint64 (GST_BUFFER_OFFSET_NONE, GST_BUFFER_OFFSET (buf));
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fail_unless_equals_uint64 (GST_BUFFER_OFFSET_NONE,
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GST_BUFFER_OFFSET_END (buf));
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gst_buffer_map (buf, &map, GST_MAP_READ);
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/* 0x08 means it is audio */
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fail_unless_equals_int (0x08, map.data[0]);
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/* timestamp should reflect the duration_ms */
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fail_unless_equals_int (duration_ms, map.data[6]);
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/* 0xb2 means Speex, 16000Hz, Mono */
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fail_unless_equals_int (0xb2, map.data[11]);
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/* verify content is intact */
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fail_unless_equals_int (0, memcmp (&map.data[12], speex_buf,
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sizeof (speex_buf)));
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gst_buffer_unmap (buf, &map);
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gst_buffer_unref (buf);
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gst_harness_teardown (h);
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}
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GST_END_TEST;
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static void
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check_buf_type_timestamp (GstBuffer * buf, gint packet_type, gint timestamp)
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{
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GstMapInfo map = GST_MAP_INFO_INIT;
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gst_buffer_map (buf, &map, GST_MAP_READ);
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fail_unless_equals_int (packet_type, map.data[0]);
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fail_unless_equals_int (timestamp, map.data[6]);
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gst_buffer_unmap (buf, &map);
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gst_buffer_unref (buf);
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}
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static const gint AUDIO = 0x08;
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static const gint VIDEO = 0x09;
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GST_START_TEST (test_increasing_timestamp_when_pts_none)
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{
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gint timestamp = 3;
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GstClockTime base_time = 42 * GST_SECOND;
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GstPad *audio_sink, *video_sink, *audio_src, *video_src;
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GstHarness *h, *audio, *video, *audio_q, *video_q;
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GstCaps *audio_caps, *video_caps;
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GstBuffer *buf;
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h = gst_harness_new_with_padnames ("flvmux", NULL, "src");
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audio = gst_harness_new_with_element (h->element, "audio", NULL);
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video = gst_harness_new_with_element (h->element, "video", NULL);
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audio_q = gst_harness_new ("queue");
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video_q = gst_harness_new ("queue");
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audio_sink = GST_PAD_PEER (audio->srcpad);
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video_sink = GST_PAD_PEER (video->srcpad);
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audio_src = GST_PAD_PEER (audio_q->sinkpad);
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video_src = GST_PAD_PEER (video_q->sinkpad);
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gst_pad_unlink (audio->srcpad, audio_sink);
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gst_pad_unlink (video->srcpad, video_sink);
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gst_pad_unlink (audio_src, audio_q->sinkpad);
|
|
gst_pad_unlink (video_src, video_q->sinkpad);
|
|
gst_pad_link (audio_src, audio_sink);
|
|
gst_pad_link (video_src, video_sink);
|
|
|
|
audio_caps = gst_caps_new_simple ("audio/x-speex",
|
|
"rate", G_TYPE_INT, 16000, "channels", G_TYPE_INT, 1, NULL);
|
|
gst_harness_set_src_caps (audio_q, audio_caps);
|
|
video_caps = gst_caps_new_simple ("video/x-h264",
|
|
"stream-format", G_TYPE_STRING, "avc", NULL);
|
|
gst_harness_set_src_caps (video_q, video_caps);
|
|
|
|
/* Push audio + video + audio with increasing DTS, but PTS for video is
|
|
* GST_CLOCK_TIME_NONE
|
|
*/
|
|
buf = gst_buffer_new ();
|
|
GST_BUFFER_DTS (buf) = timestamp * GST_MSECOND + base_time;
|
|
GST_BUFFER_PTS (buf) = timestamp * GST_MSECOND + base_time;
|
|
gst_harness_push (audio_q, buf);
|
|
|
|
buf = gst_buffer_new ();
|
|
GST_BUFFER_DTS (buf) = (timestamp + 1) * GST_MSECOND + base_time;
|
|
GST_BUFFER_PTS (buf) = GST_CLOCK_TIME_NONE;
|
|
gst_harness_push (video_q, buf);
|
|
|
|
buf = gst_buffer_new ();
|
|
GST_BUFFER_DTS (buf) = (timestamp + 2) * GST_MSECOND + base_time;
|
|
GST_BUFFER_PTS (buf) = (timestamp + 2) * GST_MSECOND + base_time;
|
|
gst_harness_push (audio_q, buf);
|
|
|
|
/* Pull two metadata packets out */
|
|
gst_buffer_unref (gst_harness_pull (h));
|
|
gst_buffer_unref (gst_harness_pull (h));
|
|
|
|
/* Check that we receive the packets in monotonically increasing order and
|
|
* that their timestamps are correct (should start at 0)
|
|
*/
|
|
buf = gst_harness_pull (h);
|
|
check_buf_type_timestamp (buf, AUDIO, 0);
|
|
buf = gst_harness_pull (h);
|
|
check_buf_type_timestamp (buf, VIDEO, 1);
|
|
|
|
/* teardown */
|
|
gst_harness_teardown (h);
|
|
gst_harness_teardown (audio);
|
|
gst_harness_teardown (video);
|
|
gst_harness_teardown (audio_q);
|
|
gst_harness_teardown (video_q);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
typedef struct
|
|
{
|
|
GstHarness *a_sink;
|
|
GstHarness *v_sink;
|
|
} DemuxHarnesses;
|
|
|
|
static void
|
|
flvdemux_pad_added (GstElement * flvdemux, GstPad * srcpad, DemuxHarnesses * h)
|
|
{
|
|
GstCaps *caps = gst_pad_get_current_caps (srcpad);
|
|
const gchar *name = gst_structure_get_name (gst_caps_get_structure (caps, 0));
|
|
|
|
if (h->a_sink && g_ascii_strncasecmp ("audio", name, 5) == 0)
|
|
gst_harness_add_element_src_pad (h->a_sink, srcpad);
|
|
else if (h->v_sink && g_ascii_strncasecmp ("video", name, 5) == 0)
|
|
gst_harness_add_element_src_pad (h->v_sink, srcpad);
|
|
else
|
|
ck_abort_msg ("Unexpected demux pad: %s", GST_STR_NULL (name));
|
|
|
|
gst_caps_unref (caps);
|
|
}
|
|
|
|
GST_START_TEST (test_video_caps_late)
|
|
{
|
|
GstHarness *mux = gst_harness_new_with_padnames ("flvmux", NULL, "src");
|
|
GstHarness *a_src =
|
|
gst_harness_new_with_element (mux->element, "audio", NULL);
|
|
GstHarness *v_src =
|
|
gst_harness_new_with_element (mux->element, "video", NULL);
|
|
GstHarness *demux = gst_harness_new_with_padnames ("flvdemux", "sink", NULL);
|
|
GstHarness *a_sink =
|
|
gst_harness_new_with_element (demux->element, NULL, NULL);
|
|
GstHarness *v_sink =
|
|
gst_harness_new_with_element (demux->element, NULL, NULL);
|
|
DemuxHarnesses harnesses = { a_sink, v_sink };
|
|
GstTestClock *tclock;
|
|
|
|
g_object_set (mux->element, "streamable", TRUE,
|
|
"latency", G_GUINT64_CONSTANT (1), NULL);
|
|
gst_harness_use_testclock (mux);
|
|
|
|
g_signal_connect (demux->element, "pad-added",
|
|
G_CALLBACK (flvdemux_pad_added), &harnesses);
|
|
gst_harness_add_sink_harness (mux, demux);
|
|
|
|
gst_harness_set_src_caps_str (a_src,
|
|
"audio/x-speex, rate=(int)16000, channels=(int)1");
|
|
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (a_src,
|
|
gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY,
|
|
speex_hdr0, sizeof (speex_hdr0), 0, sizeof (speex_hdr0), NULL,
|
|
NULL)));
|
|
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (a_src,
|
|
gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY,
|
|
speex_hdr1, sizeof (speex_hdr1), 0, sizeof (speex_hdr1), NULL,
|
|
NULL)));
|
|
|
|
/* Wait a little and make sure no clock was scheduled as this shouldn't happen
|
|
* before the caps are set */
|
|
g_usleep (40 * 1000);
|
|
tclock = gst_harness_get_testclock (mux);
|
|
fail_unless (gst_test_clock_get_next_entry_time (tclock) ==
|
|
GST_CLOCK_TIME_NONE);
|
|
|
|
gst_harness_set_src_caps_str (v_src,
|
|
"video/x-h264, stream-format=(string)avc, alignment=(string)au, "
|
|
"codec_data=(buffer)0142c00cffe1000b6742c00c95a7201e1108d401000468ce3c80");
|
|
|
|
gst_harness_crank_single_clock_wait (mux);
|
|
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (a_src,
|
|
gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY,
|
|
speex_buf, sizeof (speex_buf), 0, sizeof (speex_buf), NULL,
|
|
NULL)));
|
|
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (v_src,
|
|
gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY,
|
|
h264_buf, sizeof (h264_buf), 0, sizeof (h264_buf), NULL, NULL)));
|
|
|
|
gst_harness_crank_single_clock_wait (mux);
|
|
gst_harness_crank_single_clock_wait (mux);
|
|
gst_harness_crank_single_clock_wait (mux);
|
|
|
|
|
|
/* push from flvmux to demux */
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 6));
|
|
|
|
/* verify we got 2x audio and 1x video buffers out of flvdemux */
|
|
gst_buffer_unref (gst_harness_pull (a_sink));
|
|
gst_buffer_unref (gst_harness_pull (a_sink));
|
|
gst_buffer_unref (gst_harness_pull (v_sink));
|
|
|
|
fail_unless (gst_test_clock_get_next_entry_time (tclock) ==
|
|
GST_CLOCK_TIME_NONE);
|
|
|
|
g_clear_object (&tclock);
|
|
gst_harness_teardown (a_src);
|
|
gst_harness_teardown (v_src);
|
|
gst_harness_teardown (mux);
|
|
gst_harness_teardown (a_sink);
|
|
gst_harness_teardown (v_sink);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
guint8 raw_frame_short[] = {
|
|
0x27, 0x00, 0x03, 0x20, 0x64, 0x1c
|
|
};
|
|
|
|
GST_START_TEST (test_video_caps_change_streamable)
|
|
{
|
|
GstEvent *event;
|
|
GstCaps *a_caps1, *v_caps1, *v_caps2, *ret_caps;
|
|
GstHarness *mux = gst_harness_new_with_padnames ("flvmux", NULL, "src");
|
|
GstHarness *a_src =
|
|
gst_harness_new_with_element (mux->element, "audio", NULL);
|
|
GstHarness *v_src =
|
|
gst_harness_new_with_element (mux->element, "video", NULL);
|
|
GstHarness *demux = gst_harness_new_with_padnames ("flvdemux", "sink", NULL);
|
|
GstHarness *a_sink =
|
|
gst_harness_new_with_element (demux->element, NULL, NULL);
|
|
GstHarness *v_sink =
|
|
gst_harness_new_with_element (demux->element, NULL, NULL);
|
|
DemuxHarnesses harnesses = { a_sink, v_sink };
|
|
|
|
const GstClockTime base_time = 123456789;
|
|
const GstClockTime duration_ms = 20;
|
|
const GstClockTime duration = duration_ms * GST_MSECOND;
|
|
|
|
g_object_set (mux->element, "streamable", TRUE, NULL);
|
|
|
|
g_signal_connect (demux->element, "pad-added",
|
|
G_CALLBACK (flvdemux_pad_added), &harnesses);
|
|
gst_harness_add_sink_harness (mux, demux);
|
|
|
|
a_caps1 =
|
|
gst_caps_from_string
|
|
("audio/mpeg, mpegversion=(int)4, framed=(boolean)true, stream-format=(string)raw, "
|
|
"rate=(int)44100, channels=(int)1, codec_data=(buffer)1208");
|
|
|
|
v_caps1 = gst_caps_from_string ("video/x-h264, stream-format=(string)avc, "
|
|
"codec_data=(buffer)0142c00cffe1000b6742c00c95a7201e1108d401000468ce3c80");
|
|
|
|
gst_harness_set_src_caps (a_src, gst_caps_ref (a_caps1));
|
|
gst_harness_set_src_caps (v_src, gst_caps_ref (v_caps1));
|
|
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (a_src,
|
|
create_buffer (raw_frame_short, sizeof (raw_frame_short), base_time,
|
|
duration)));
|
|
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (v_src,
|
|
create_buffer (h264_buf, sizeof (h264_buf), base_time, duration)));
|
|
|
|
gst_harness_crank_single_clock_wait (mux);
|
|
|
|
/* push from flvmux to demux */
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 6));
|
|
|
|
/* should accept without the constraint */
|
|
while ((event = gst_harness_try_pull_event (v_sink))) {
|
|
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
|
|
gst_event_parse_caps (event, &ret_caps);
|
|
GST_LOG ("v_caps1 %" GST_PTR_FORMAT ", ret caps %" GST_PTR_FORMAT,
|
|
v_caps1, ret_caps);
|
|
fail_unless (gst_caps_is_equal (v_caps1, ret_caps));
|
|
}
|
|
gst_event_unref (event);
|
|
}
|
|
|
|
/* caps change */
|
|
v_caps2 = gst_caps_from_string ("video/x-h264, stream-format=(string)avc, "
|
|
"codec_data=(buffer)0164001fffe1001c6764001facd9405005bb016a02020280000003008000001e478c18cb01000568ebecb22c");
|
|
|
|
gst_harness_set_src_caps (v_src, gst_caps_ref (v_caps2));
|
|
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (v_src,
|
|
create_buffer (h264_buf, sizeof (h264_buf), base_time + duration,
|
|
duration)));
|
|
|
|
gst_harness_crank_single_clock_wait (mux);
|
|
|
|
/* push from flvmux to demux */
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 2));
|
|
|
|
/* should accept without the constraint */
|
|
while ((event = gst_harness_try_pull_event (v_sink))) {
|
|
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
|
|
gst_event_parse_caps (event, &ret_caps);
|
|
GST_LOG ("v_caps2 %" GST_PTR_FORMAT ", ret caps %" GST_PTR_FORMAT,
|
|
v_caps2, ret_caps);
|
|
fail_unless (gst_caps_is_equal (v_caps2, ret_caps));
|
|
}
|
|
gst_event_unref (event);
|
|
}
|
|
|
|
/* verify we got 1x audio and 2x video buffers out of flvdemux */
|
|
gst_buffer_unref (gst_harness_pull (a_sink));
|
|
gst_buffer_unref (gst_harness_pull (v_sink));
|
|
gst_buffer_unref (gst_harness_pull (v_sink));
|
|
gst_caps_unref (a_caps1);
|
|
gst_caps_unref (v_caps1);
|
|
gst_caps_unref (v_caps2);
|
|
|
|
gst_harness_teardown (a_src);
|
|
gst_harness_teardown (v_src);
|
|
gst_harness_teardown (mux);
|
|
gst_harness_teardown (a_sink);
|
|
gst_harness_teardown (v_sink);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_audio_caps_change_streamable)
|
|
{
|
|
GstEvent *event;
|
|
GstCaps *a_caps1, *a_caps2, *v_caps1, *ret_caps;
|
|
GstHarness *mux = gst_harness_new_with_padnames ("flvmux", NULL, "src");
|
|
GstHarness *a_src =
|
|
gst_harness_new_with_element (mux->element, "audio", NULL);
|
|
GstHarness *v_src =
|
|
gst_harness_new_with_element (mux->element, "video", NULL);
|
|
GstHarness *demux = gst_harness_new_with_padnames ("flvdemux", "sink", NULL);
|
|
GstHarness *a_sink =
|
|
gst_harness_new_with_element (demux->element, NULL, NULL);
|
|
GstHarness *v_sink =
|
|
gst_harness_new_with_element (demux->element, NULL, NULL);
|
|
DemuxHarnesses harnesses = { a_sink, v_sink };
|
|
|
|
const GstClockTime base_time = 123456789;
|
|
const GstClockTime duration_ms = 20;
|
|
const GstClockTime duration = duration_ms * GST_MSECOND;
|
|
|
|
g_object_set (mux->element, "streamable", TRUE, NULL);
|
|
|
|
g_signal_connect (demux->element, "pad-added",
|
|
G_CALLBACK (flvdemux_pad_added), &harnesses);
|
|
gst_harness_add_sink_harness (mux, demux);
|
|
|
|
a_caps1 =
|
|
gst_caps_from_string
|
|
("audio/mpeg, mpegversion=(int)4, framed=(boolean)true, stream-format=(string)raw, "
|
|
"rate=(int)44100, channels=(int)1, codec_data=(buffer)1208");
|
|
|
|
v_caps1 = gst_caps_from_string ("video/x-h264, stream-format=(string)avc, "
|
|
"codec_data=(buffer)0142c00cffe1000b6742c00c95a7201e1108d401000468ce3c80");
|
|
|
|
gst_harness_set_src_caps (a_src, gst_caps_ref (a_caps1));
|
|
gst_harness_set_src_caps (v_src, gst_caps_ref (v_caps1));
|
|
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (a_src,
|
|
create_buffer (raw_frame_short, sizeof (raw_frame_short), base_time,
|
|
duration)));
|
|
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (v_src,
|
|
create_buffer (h264_buf, sizeof (h264_buf), base_time, duration)));
|
|
|
|
gst_harness_crank_single_clock_wait (mux);
|
|
|
|
/* push from flvmux to demux */
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 6));
|
|
|
|
/* should accept without the constraint */
|
|
while ((event = gst_harness_try_pull_event (a_sink))) {
|
|
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
|
|
gst_event_parse_caps (event, &ret_caps);
|
|
GST_LOG ("a_caps1 %" GST_PTR_FORMAT ", ret caps %" GST_PTR_FORMAT,
|
|
a_caps1, ret_caps);
|
|
fail_unless (gst_caps_is_equal (a_caps1, ret_caps));
|
|
}
|
|
gst_event_unref (event);
|
|
}
|
|
|
|
/* caps change */
|
|
a_caps2 =
|
|
gst_caps_from_string
|
|
("audio/mpeg, mpegversion=(int)4, framed=(boolean)true, stream-format=(string)raw, "
|
|
"rate=(int)22050, channels=(int)1, codec_data=(buffer)1388");
|
|
|
|
gst_harness_set_src_caps (a_src, gst_caps_ref (a_caps2));
|
|
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (a_src,
|
|
create_buffer (raw_frame_short, sizeof (raw_frame_short),
|
|
base_time + duration, duration)));
|
|
|
|
gst_harness_crank_single_clock_wait (mux);
|
|
|
|
/* push from flvmux to demux */
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 2));
|
|
|
|
/* should accept without the constraint */
|
|
while ((event = gst_harness_try_pull_event (a_sink))) {
|
|
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
|
|
gst_event_parse_caps (event, &ret_caps);
|
|
GST_LOG ("a_caps2 %" GST_PTR_FORMAT ", ret caps %" GST_PTR_FORMAT,
|
|
a_caps2, ret_caps);
|
|
fail_unless (gst_caps_is_equal (a_caps2, ret_caps));
|
|
}
|
|
gst_event_unref (event);
|
|
}
|
|
|
|
/* verify we got 2x audio and 1x video buffers out of flvdemux */
|
|
gst_buffer_unref (gst_harness_pull (a_sink));
|
|
gst_buffer_unref (gst_harness_pull (a_sink));
|
|
gst_buffer_unref (gst_harness_pull (v_sink));
|
|
gst_caps_unref (a_caps1);
|
|
gst_caps_unref (a_caps2);
|
|
gst_caps_unref (v_caps1);
|
|
|
|
gst_harness_teardown (a_src);
|
|
gst_harness_teardown (v_src);
|
|
gst_harness_teardown (mux);
|
|
gst_harness_teardown (a_sink);
|
|
gst_harness_teardown (v_sink);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_video_caps_change_streamable_single)
|
|
{
|
|
GstEvent *event;
|
|
GstCaps *v_caps1, *v_caps2, *ret_caps;
|
|
GstHarness *mux = gst_harness_new_with_padnames ("flvmux", NULL, "src");
|
|
GstHarness *v_src =
|
|
gst_harness_new_with_element (mux->element, "video", NULL);
|
|
GstHarness *demux = gst_harness_new_with_padnames ("flvdemux", "sink", NULL);
|
|
GstHarness *v_sink =
|
|
gst_harness_new_with_element (demux->element, NULL, NULL);
|
|
DemuxHarnesses harnesses = { NULL, v_sink };
|
|
|
|
const GstClockTime base_time = 123456789;
|
|
const GstClockTime duration_ms = 20;
|
|
const GstClockTime duration = duration_ms * GST_MSECOND;
|
|
|
|
g_object_set (mux->element, "streamable", TRUE, NULL);
|
|
|
|
g_signal_connect (demux->element, "pad-added",
|
|
G_CALLBACK (flvdemux_pad_added), &harnesses);
|
|
gst_harness_add_sink_harness (mux, demux);
|
|
|
|
v_caps1 = gst_caps_from_string ("video/x-h264, stream-format=(string)avc, "
|
|
"codec_data=(buffer)0142c00cffe1000b6742c00c95a7201e1108d401000468ce3c80");
|
|
|
|
gst_harness_set_src_caps (v_src, gst_caps_ref (v_caps1));
|
|
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (v_src,
|
|
create_buffer (h264_buf, sizeof (h264_buf), base_time, duration)));
|
|
|
|
/* push from flvmux to demux */
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 4));
|
|
|
|
/* should accept without the constraint */
|
|
while ((event = gst_harness_try_pull_event (v_sink))) {
|
|
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
|
|
gst_event_parse_caps (event, &ret_caps);
|
|
GST_LOG ("v_caps1 %" GST_PTR_FORMAT ", ret caps %" GST_PTR_FORMAT,
|
|
v_caps1, ret_caps);
|
|
fail_unless (gst_caps_is_equal (v_caps1, ret_caps));
|
|
}
|
|
gst_event_unref (event);
|
|
}
|
|
|
|
/* caps change */
|
|
v_caps2 = gst_caps_from_string ("video/x-h264, stream-format=(string)avc, "
|
|
"codec_data=(buffer)0164001fffe1001c6764001facd9405005bb016a02020280000003008000001e478c18cb01000568ebecb22c");
|
|
|
|
gst_harness_set_src_caps (v_src, gst_caps_ref (v_caps2));
|
|
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (v_src,
|
|
create_buffer (h264_buf, sizeof (h264_buf), base_time + duration,
|
|
duration)));
|
|
|
|
/* push from flvmux to demux */
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 2));
|
|
|
|
/* should accept without the constraint */
|
|
while ((event = gst_harness_try_pull_event (v_sink))) {
|
|
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
|
|
gst_event_parse_caps (event, &ret_caps);
|
|
GST_LOG ("v_caps2 %" GST_PTR_FORMAT ", ret caps %" GST_PTR_FORMAT,
|
|
v_caps2, ret_caps);
|
|
fail_unless (gst_caps_is_equal (v_caps2, ret_caps));
|
|
}
|
|
gst_event_unref (event);
|
|
}
|
|
|
|
/* verify we got 2x video buffers out of flvdemux */
|
|
gst_buffer_unref (gst_harness_pull (v_sink));
|
|
gst_buffer_unref (gst_harness_pull (v_sink));
|
|
gst_caps_unref (v_caps1);
|
|
gst_caps_unref (v_caps2);
|
|
|
|
gst_harness_teardown (v_src);
|
|
gst_harness_teardown (mux);
|
|
gst_harness_teardown (v_sink);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_audio_caps_change_streamable_single)
|
|
{
|
|
GstEvent *event;
|
|
GstCaps *a_caps1, *a_caps2, *ret_caps;
|
|
GstHarness *mux = gst_harness_new_with_padnames ("flvmux", NULL, "src");
|
|
GstHarness *a_src =
|
|
gst_harness_new_with_element (mux->element, "audio", NULL);
|
|
GstHarness *demux = gst_harness_new_with_padnames ("flvdemux", "sink", NULL);
|
|
GstHarness *a_sink =
|
|
gst_harness_new_with_element (demux->element, NULL, NULL);
|
|
DemuxHarnesses harnesses = { a_sink, NULL };
|
|
|
|
const GstClockTime base_time = 123456789;
|
|
const GstClockTime duration_ms = 20;
|
|
const GstClockTime duration = duration_ms * GST_MSECOND;
|
|
|
|
g_object_set (mux->element, "streamable", TRUE, NULL);
|
|
|
|
g_signal_connect (demux->element, "pad-added",
|
|
G_CALLBACK (flvdemux_pad_added), &harnesses);
|
|
gst_harness_add_sink_harness (mux, demux);
|
|
|
|
a_caps1 =
|
|
gst_caps_from_string
|
|
("audio/mpeg, mpegversion=(int)4, framed=(boolean)true, stream-format=(string)raw, "
|
|
"rate=(int)44100, channels=(int)1, codec_data=(buffer)1208");
|
|
|
|
gst_harness_set_src_caps (a_src, gst_caps_ref (a_caps1));
|
|
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (a_src,
|
|
create_buffer (raw_frame_short, sizeof (raw_frame_short), base_time,
|
|
duration)));
|
|
|
|
/* push from flvmux to demux */
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 4));
|
|
|
|
/* should accept without the constraint */
|
|
while ((event = gst_harness_try_pull_event (a_sink))) {
|
|
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
|
|
gst_event_parse_caps (event, &ret_caps);
|
|
GST_LOG ("a_caps1 %" GST_PTR_FORMAT ", ret caps %" GST_PTR_FORMAT,
|
|
a_caps1, ret_caps);
|
|
fail_unless (gst_caps_is_equal (a_caps1, ret_caps));
|
|
}
|
|
gst_event_unref (event);
|
|
}
|
|
|
|
/* caps change */
|
|
a_caps2 =
|
|
gst_caps_from_string
|
|
("audio/mpeg, mpegversion=(int)4, framed=(boolean)true, stream-format=(string)raw, "
|
|
"rate=(int)22050, channels=(int)1, codec_data=(buffer)1388");
|
|
|
|
gst_harness_set_src_caps (a_src, gst_caps_ref (a_caps2));
|
|
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (a_src,
|
|
create_buffer (raw_frame_short, sizeof (raw_frame_short),
|
|
base_time + duration, duration)));
|
|
|
|
/* push from flvmux to demux */
|
|
fail_unless_equals_int (GST_FLOW_OK, gst_harness_sink_push_many (mux, 2));
|
|
|
|
/* should accept without the constraint */
|
|
while ((event = gst_harness_try_pull_event (a_sink))) {
|
|
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
|
|
gst_event_parse_caps (event, &ret_caps);
|
|
GST_LOG ("a_caps2 %" GST_PTR_FORMAT ", ret caps %" GST_PTR_FORMAT,
|
|
a_caps2, ret_caps);
|
|
fail_unless (gst_caps_is_equal (a_caps2, ret_caps));
|
|
}
|
|
gst_event_unref (event);
|
|
}
|
|
|
|
/* verify we got 2x audio out of flvdemux */
|
|
gst_buffer_unref (gst_harness_pull (a_sink));
|
|
gst_buffer_unref (gst_harness_pull (a_sink));
|
|
gst_caps_unref (a_caps1);
|
|
gst_caps_unref (a_caps2);
|
|
|
|
gst_harness_teardown (a_src);
|
|
gst_harness_teardown (mux);
|
|
gst_harness_teardown (a_sink);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
typedef struct
|
|
{
|
|
guint media_type;
|
|
guint64 ts; /* timestamp in ms */
|
|
} InputData;
|
|
|
|
GST_START_TEST (test_incrementing_timestamps)
|
|
{
|
|
GstPad *audio_sink, *video_sink, *audio_src, *video_src;
|
|
GstHarness *h, *audio, *video, *audio_q, *video_q;
|
|
guint i;
|
|
GstEvent *event;
|
|
guint32 prev_pts;
|
|
InputData input[] = {
|
|
{AUDIO, 155},
|
|
{VIDEO, 156},
|
|
{VIDEO, 190},
|
|
{AUDIO, 176},
|
|
{AUDIO, 197},
|
|
};
|
|
|
|
/* setup flvmuxer with queues in front */
|
|
h = gst_harness_new_with_padnames ("flvmux", NULL, "src");
|
|
audio = gst_harness_new_with_element (h->element, "audio", NULL);
|
|
video = gst_harness_new_with_element (h->element, "video", NULL);
|
|
audio_q = gst_harness_new ("queue");
|
|
video_q = gst_harness_new ("queue");
|
|
audio_sink = GST_PAD_PEER (audio->srcpad);
|
|
video_sink = GST_PAD_PEER (video->srcpad);
|
|
audio_src = GST_PAD_PEER (audio_q->sinkpad);
|
|
video_src = GST_PAD_PEER (video_q->sinkpad);
|
|
gst_pad_unlink (audio->srcpad, audio_sink);
|
|
gst_pad_unlink (video->srcpad, video_sink);
|
|
gst_pad_unlink (audio_src, audio_q->sinkpad);
|
|
gst_pad_unlink (video_src, video_q->sinkpad);
|
|
gst_pad_link (audio_src, audio_sink);
|
|
gst_pad_link (video_src, video_sink);
|
|
g_object_set (h->element, "streamable", TRUE, NULL);
|
|
|
|
gst_harness_set_src_caps_str (audio_q,
|
|
"audio/mpeg, mpegversion=(int)4, "
|
|
"rate=(int)44100, channels=(int)1, "
|
|
"stream-format=(string)raw, codec_data=(buffer)1208");
|
|
|
|
gst_harness_set_src_caps_str (video_q,
|
|
"video/x-h264, stream-format=(string)avc, alignment=(string)au, "
|
|
"codec_data=(buffer)0142c00dffe1000d6742c00d95a0507c807844235001000468ce3c80");
|
|
|
|
for (i = 0; i < G_N_ELEMENTS (input); i++) {
|
|
InputData *d = &input[i];
|
|
GstBuffer *buf = gst_buffer_new ();
|
|
|
|
GST_BUFFER_DTS (buf) = GST_BUFFER_PTS (buf) = d->ts * GST_MSECOND;
|
|
|
|
if (d->media_type == AUDIO)
|
|
gst_harness_push (audio_q, buf);
|
|
else
|
|
gst_harness_push (video_q, buf);
|
|
}
|
|
|
|
gst_harness_push_event (audio_q, gst_event_new_eos ());
|
|
gst_harness_push_event (video_q, gst_event_new_eos ());
|
|
|
|
while ((event = gst_harness_pull_event (h)) != NULL) {
|
|
GstEventType event_type = GST_EVENT_TYPE (event);
|
|
gst_event_unref (event);
|
|
|
|
if (event_type == GST_EVENT_EOS)
|
|
break;
|
|
}
|
|
|
|
/* pull the flv metadata */
|
|
gst_buffer_unref (gst_harness_pull (h));
|
|
gst_buffer_unref (gst_harness_pull (h));
|
|
gst_buffer_unref (gst_harness_pull (h));
|
|
|
|
/* verify pts in the flvheader is increasing */
|
|
prev_pts = 0;
|
|
for (i = 0; i < G_N_ELEMENTS (input); i++) {
|
|
GstBuffer *buf = gst_harness_pull (h);
|
|
GstMapInfo map;
|
|
guint32 pts;
|
|
|
|
fail_unless (buf != NULL);
|
|
|
|
gst_buffer_map (buf, &map, GST_MAP_READ);
|
|
pts = GST_READ_UINT24_BE (map.data + 4);
|
|
GST_DEBUG ("media=%u, pts = %u\n", map.data[0], pts);
|
|
fail_unless (pts >= prev_pts);
|
|
prev_pts = pts;
|
|
gst_buffer_unmap (buf, &map);
|
|
gst_buffer_unref (buf);
|
|
}
|
|
|
|
/* teardown */
|
|
gst_harness_teardown (h);
|
|
gst_harness_teardown (audio);
|
|
gst_harness_teardown (video);
|
|
gst_harness_teardown (audio_q);
|
|
gst_harness_teardown (video_q);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_rollover_timestamps)
|
|
{
|
|
GstPad *audio_sink, *video_sink, *audio_src, *video_src;
|
|
GstHarness *h, *audio, *video, *audio_q, *video_q;
|
|
GstEvent *event;
|
|
guint i;
|
|
guint64 rollover_pts = (guint64) G_MAXUINT32 + 100;
|
|
InputData input[] = {
|
|
{AUDIO, 0}
|
|
,
|
|
{VIDEO, 0}
|
|
,
|
|
{VIDEO, (guint64) G_MAXUINT32 - 100}
|
|
,
|
|
{AUDIO, (guint64) G_MAXUINT32 - 95}
|
|
,
|
|
{AUDIO, rollover_pts}
|
|
,
|
|
};
|
|
|
|
/* setup flvmuxer with queues in front */
|
|
h = gst_harness_new_with_padnames ("flvmux", NULL, "src");
|
|
audio = gst_harness_new_with_element (h->element, "audio", NULL);
|
|
video = gst_harness_new_with_element (h->element, "video", NULL);
|
|
audio_q = gst_harness_new ("queue");
|
|
video_q = gst_harness_new ("queue");
|
|
audio_sink = GST_PAD_PEER (audio->srcpad);
|
|
video_sink = GST_PAD_PEER (video->srcpad);
|
|
audio_src = GST_PAD_PEER (audio_q->sinkpad);
|
|
video_src = GST_PAD_PEER (video_q->sinkpad);
|
|
gst_pad_unlink (audio->srcpad, audio_sink);
|
|
gst_pad_unlink (video->srcpad, video_sink);
|
|
gst_pad_unlink (audio_src, audio_q->sinkpad);
|
|
gst_pad_unlink (video_src, video_q->sinkpad);
|
|
gst_pad_link (audio_src, audio_sink);
|
|
gst_pad_link (video_src, video_sink);
|
|
g_object_set (h->element, "streamable", TRUE, NULL);
|
|
|
|
gst_harness_set_src_caps_str (audio_q,
|
|
"audio/mpeg, mpegversion=(int)4, "
|
|
"rate=(int)44100, channels=(int)1, "
|
|
"stream-format=(string)raw, codec_data=(buffer)1208");
|
|
|
|
gst_harness_set_src_caps_str (video_q,
|
|
"video/x-h264, stream-format=(string)avc, alignment=(string)au, "
|
|
"codec_data=(buffer)0142c00dffe1000d6742c00d95a0507c807844235001000468ce3c80");
|
|
|
|
for (i = 0; i < G_N_ELEMENTS (input); i++) {
|
|
InputData *d = &input[i];
|
|
GstBuffer *buf = gst_buffer_new ();
|
|
|
|
GST_BUFFER_DTS (buf) = GST_BUFFER_PTS (buf) = d->ts * GST_MSECOND;
|
|
GST_DEBUG ("Push media=%u, pts=%" G_GUINT64_FORMAT " (%" GST_TIME_FORMAT
|
|
")", d->media_type, d->ts, GST_TIME_ARGS (GST_BUFFER_PTS (buf)));
|
|
|
|
if (d->media_type == AUDIO)
|
|
gst_harness_push (audio_q, buf);
|
|
else
|
|
gst_harness_push (video_q, buf);
|
|
|
|
}
|
|
gst_harness_push_event (audio_q, gst_event_new_eos ());
|
|
gst_harness_push_event (video_q, gst_event_new_eos ());
|
|
|
|
while ((event = gst_harness_pull_event (h)) != NULL) {
|
|
GstEventType event_type = GST_EVENT_TYPE (event);
|
|
gst_event_unref (event);
|
|
|
|
if (event_type == GST_EVENT_EOS)
|
|
break;
|
|
}
|
|
|
|
/* pull the flv metadata */
|
|
gst_buffer_unref (gst_harness_pull (h));
|
|
gst_buffer_unref (gst_harness_pull (h));
|
|
gst_buffer_unref (gst_harness_pull (h));
|
|
gst_buffer_unref (gst_harness_pull (h));
|
|
|
|
/* verify rollover pts in the flvheader is handled */
|
|
for (i = 0; i < G_N_ELEMENTS (input); i++) {
|
|
GstBuffer *buf = gst_harness_pull (h);
|
|
GstMapInfo map;
|
|
guint32 pts, pts_ext;
|
|
|
|
fail_unless (buf != NULL);
|
|
|
|
gst_buffer_map (buf, &map, GST_MAP_READ);
|
|
pts = GST_READ_UINT24_BE (map.data + 4);
|
|
pts_ext = GST_READ_UINT8 (map.data + 7);
|
|
pts |= pts_ext << 24;
|
|
GST_DEBUG ("media=%u, pts=%u (%" GST_TIME_FORMAT ")",
|
|
map.data[0], pts, GST_TIME_ARGS (pts * GST_MSECOND));
|
|
fail_unless (pts == (guint32) input[i].ts);
|
|
gst_buffer_unmap (buf, &map);
|
|
gst_buffer_unref (buf);
|
|
}
|
|
|
|
/* teardown */
|
|
gst_harness_teardown (h);
|
|
gst_harness_teardown (audio);
|
|
gst_harness_teardown (video);
|
|
gst_harness_teardown (audio_q);
|
|
gst_harness_teardown (video_q);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static Suite *
|
|
flvmux_suite (void)
|
|
{
|
|
Suite *s = suite_create ("flvmux");
|
|
TCase *tc_chain = tcase_create ("general");
|
|
gint loop = 16;
|
|
|
|
suite_add_tcase (s, tc_chain);
|
|
|
|
#ifdef HAVE_VALGRIND
|
|
if (RUNNING_ON_VALGRIND) {
|
|
loop = 1;
|
|
}
|
|
#endif
|
|
|
|
tcase_add_loop_test (tc_chain, test_index_writing, 0, loop);
|
|
|
|
tcase_add_test (tc_chain, test_speex_streamable);
|
|
tcase_add_test (tc_chain, test_increasing_timestamp_when_pts_none);
|
|
tcase_add_test (tc_chain, test_video_caps_late);
|
|
tcase_add_test (tc_chain, test_audio_caps_change_streamable);
|
|
tcase_add_test (tc_chain, test_video_caps_change_streamable);
|
|
tcase_add_test (tc_chain, test_audio_caps_change_streamable_single);
|
|
tcase_add_test (tc_chain, test_video_caps_change_streamable_single);
|
|
tcase_add_test (tc_chain, test_incrementing_timestamps);
|
|
tcase_add_test (tc_chain, test_rollover_timestamps);
|
|
|
|
return s;
|
|
}
|
|
|
|
GST_CHECK_MAIN (flvmux)
|