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153406eef5
Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-sections.txt: * docs/plugins/gst-plugins-base-plugins.args: * docs/plugins/gst-plugins-base-plugins.hierarchy: * docs/plugins/gst-plugins-base-plugins.interfaces: * docs/plugins/gst-plugins-base-plugins.prerequisites: * docs/plugins/inspect/plugin-adder.xml: * docs/plugins/inspect/plugin-alsa.xml: * docs/plugins/inspect/plugin-audioconvert.xml: * docs/plugins/inspect/plugin-audiorate.xml: * docs/plugins/inspect/plugin-audioresample.xml: * docs/plugins/inspect/plugin-audiotestsrc.xml: * docs/plugins/inspect/plugin-cdparanoia.xml: * docs/plugins/inspect/plugin-decodebin.xml: * docs/plugins/inspect/plugin-ffmpegcolorspace.xml: * docs/plugins/inspect/plugin-gdp.xml: * docs/plugins/inspect/plugin-gio.xml: * docs/plugins/inspect/plugin-gnomevfs.xml: * docs/plugins/inspect/plugin-libvisual.xml: * docs/plugins/inspect/plugin-ogg.xml: * docs/plugins/inspect/plugin-pango.xml: * docs/plugins/inspect/plugin-playback.xml: * docs/plugins/inspect/plugin-queue2.xml: * docs/plugins/inspect/plugin-subparse.xml: * docs/plugins/inspect/plugin-tcp.xml: * docs/plugins/inspect/plugin-theora.xml: * docs/plugins/inspect/plugin-typefindfunctions.xml: * docs/plugins/inspect/plugin-uridecodebin.xml: * docs/plugins/inspect/plugin-video4linux.xml: * docs/plugins/inspect/plugin-videorate.xml: * docs/plugins/inspect/plugin-videoscale.xml: * docs/plugins/inspect/plugin-videotestsrc.xml: * docs/plugins/inspect/plugin-volume.xml: * docs/plugins/inspect/plugin-vorbis.xml: * docs/plugins/inspect/plugin-ximagesink.xml: * docs/plugins/inspect/plugin-xvimagesink.xml: * gst/speexresample/gstspeexresample.c: (plugin_init): * gst/speexresample/Makefile.am: * tests/check/Makefile.am: * tests/check/elements/speexresample.c: (setup_speexresample), (GST_START_TEST), (test_pipeline): Rename the moved speexresample to audioresample, integrate into the build system and remove the old audioresample from the build system. Fixes bug #558124, #385061, #346218, #116051.
700 lines
22 KiB
C
700 lines
22 KiB
C
/* GStreamer
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*
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* unit test for speexresample, based on the audioresample unit test
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*
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* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
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* Copyright (C) <2006> Tim-Philipp Müller <tim at centricular net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <unistd.h>
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#include <gst/check/gstcheck.h>
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#include <gst/audio/audio.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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#define RESAMPLE_CAPS_FLOAT \
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"audio/x-raw-float, " \
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"channels = (int) [ 1, MAX ], " \
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"rate = (int) [ 1, MAX ], " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) { 32, 64 }"
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#define RESAMPLE_CAPS_INT \
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"audio/x-raw-int, " \
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"channels = (int) [ 1, MAX ], " \
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"rate = (int) [ 1, MAX ], " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) 16, " \
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"depth = (int) 16, " \
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"signed = (bool) TRUE"
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#define RESAMPLE_CAPS_TEMPLATE_STRING \
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RESAMPLE_CAPS_FLOAT " ; " \
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RESAMPLE_CAPS_INT
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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 (RESAMPLE_CAPS_TEMPLATE_STRING)
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);
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static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (RESAMPLE_CAPS_TEMPLATE_STRING)
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);
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static GstElement *
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setup_speexresample (int channels, int inrate, int outrate, int width,
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gboolean fp)
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{
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GstElement *speexresample;
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GstCaps *caps;
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GstStructure *structure;
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GST_DEBUG ("setup_speexresample");
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speexresample = gst_check_setup_element ("audioresample");
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if (fp)
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caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
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else
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caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_set (structure, "channels", G_TYPE_INT, channels,
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"rate", G_TYPE_INT, inrate, "width", G_TYPE_INT, width, NULL);
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if (!fp)
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gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
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fail_unless (gst_caps_is_fixed (caps));
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fail_unless (gst_element_set_state (speexresample,
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GST_STATE_PAUSED) == GST_STATE_CHANGE_SUCCESS,
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"could not set to paused");
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mysrcpad = gst_check_setup_src_pad (speexresample, &srctemplate, caps);
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gst_pad_set_caps (mysrcpad, caps);
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gst_caps_unref (caps);
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if (fp)
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caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
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else
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caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_set (structure, "channels", G_TYPE_INT, channels,
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"rate", G_TYPE_INT, outrate, "width", G_TYPE_INT, width, NULL);
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if (!fp)
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gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
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fail_unless (gst_caps_is_fixed (caps));
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mysinkpad = gst_check_setup_sink_pad (speexresample, &sinktemplate, caps);
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/* this installs a getcaps func that will always return the caps we set
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* later */
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gst_pad_set_caps (mysinkpad, caps);
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gst_pad_use_fixed_caps (mysinkpad);
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gst_pad_set_active (mysinkpad, TRUE);
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gst_pad_set_active (mysrcpad, TRUE);
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gst_caps_unref (caps);
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return speexresample;
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}
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static void
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cleanup_speexresample (GstElement * speexresample)
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{
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GST_DEBUG ("cleanup_speexresample");
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fail_unless (gst_element_set_state (speexresample,
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GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to 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 (speexresample);
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gst_check_teardown_sink_pad (speexresample);
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gst_check_teardown_element (speexresample);
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}
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static void
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fail_unless_perfect_stream (void)
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{
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guint64 timestamp = 0L, duration = 0L;
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guint64 offset = 0L, offset_end = 0L;
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GList *l;
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GstBuffer *buffer;
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for (l = buffers; l; l = l->next) {
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buffer = GST_BUFFER (l->data);
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ASSERT_BUFFER_REFCOUNT (buffer, "buffer", 1);
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GST_DEBUG ("buffer timestamp %" G_GUINT64_FORMAT ", duration %"
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G_GUINT64_FORMAT " offset %" G_GUINT64_FORMAT " offset_end %"
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G_GUINT64_FORMAT,
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GST_BUFFER_TIMESTAMP (buffer),
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GST_BUFFER_DURATION (buffer),
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GST_BUFFER_OFFSET (buffer), GST_BUFFER_OFFSET_END (buffer));
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fail_unless_equals_uint64 (timestamp, GST_BUFFER_TIMESTAMP (buffer));
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fail_unless_equals_uint64 (offset, GST_BUFFER_OFFSET (buffer));
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duration = GST_BUFFER_DURATION (buffer);
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offset_end = GST_BUFFER_OFFSET_END (buffer);
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timestamp += duration;
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offset = offset_end;
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gst_buffer_unref (buffer);
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}
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g_list_free (buffers);
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buffers = NULL;
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}
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/* this tests that the output is a perfect stream if the input is */
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static void
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test_perfect_stream_instance (int inrate, int outrate, int samples,
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int numbuffers)
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{
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GstElement *speexresample;
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GstBuffer *inbuffer, *outbuffer;
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GstCaps *caps;
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guint64 offset = 0;
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int i, j;
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gint16 *p;
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speexresample = setup_speexresample (2, inrate, outrate, 16, FALSE);
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caps = gst_pad_get_negotiated_caps (mysrcpad);
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fail_unless (gst_caps_is_fixed (caps));
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fail_unless (gst_element_set_state (speexresample,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
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"could not set to playing");
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for (j = 1; j <= numbuffers; ++j) {
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inbuffer = gst_buffer_new_and_alloc (samples * 4);
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GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (samples, inrate);
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GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1);
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GST_BUFFER_OFFSET (inbuffer) = offset;
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offset += samples;
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GST_BUFFER_OFFSET_END (inbuffer) = offset;
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gst_buffer_set_caps (inbuffer, caps);
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p = (gint16 *) GST_BUFFER_DATA (inbuffer);
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/* create a 16 bit signed ramp */
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for (i = 0; i < samples; ++i) {
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*p = -32767 + i * (65535 / samples);
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++p;
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*p = -32767 + i * (65535 / samples);
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++p;
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}
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/* pushing gives away my reference ... */
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fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
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/* ... but it ends up being collected on the global buffer list */
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fail_unless_equals_int (g_list_length (buffers), j);
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}
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/* FIXME: we should make speexresample handle eos by flushing out the last
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* samples, which will give us one more, small, buffer */
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fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
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ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
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fail_unless_perfect_stream ();
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/* cleanup */
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gst_caps_unref (caps);
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cleanup_speexresample (speexresample);
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}
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/* make sure that outgoing buffers are contiguous in timestamp/duration and
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* offset/offsetend
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*/
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GST_START_TEST (test_perfect_stream)
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{
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/* integral scalings */
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test_perfect_stream_instance (48000, 24000, 500, 20);
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test_perfect_stream_instance (48000, 12000, 500, 20);
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test_perfect_stream_instance (12000, 24000, 500, 20);
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test_perfect_stream_instance (12000, 48000, 500, 20);
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/* non-integral scalings */
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test_perfect_stream_instance (44100, 8000, 500, 20);
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test_perfect_stream_instance (8000, 44100, 500, 20);
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/* wacky scalings */
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test_perfect_stream_instance (12345, 54321, 500, 20);
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test_perfect_stream_instance (101, 99, 500, 20);
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}
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GST_END_TEST;
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/* this tests that the output is a correct discontinuous stream
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* if the input is; ie input drops in time come out the same way */
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static void
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test_discont_stream_instance (int inrate, int outrate, int samples,
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int numbuffers)
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{
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GstElement *speexresample;
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GstBuffer *inbuffer, *outbuffer;
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GstCaps *caps;
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GstClockTime ints;
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int i, j;
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gint16 *p;
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GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d",
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inrate, outrate, samples, numbuffers);
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speexresample = setup_speexresample (2, inrate, outrate, 16, FALSE);
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caps = gst_pad_get_negotiated_caps (mysrcpad);
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fail_unless (gst_caps_is_fixed (caps));
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fail_unless (gst_element_set_state (speexresample,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
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"could not set to playing");
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for (j = 1; j <= numbuffers; ++j) {
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inbuffer = gst_buffer_new_and_alloc (samples * 4);
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GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate;
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/* "drop" half the buffers */
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ints = GST_BUFFER_DURATION (inbuffer) * 2 * (j - 1);
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GST_BUFFER_TIMESTAMP (inbuffer) = ints;
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GST_BUFFER_OFFSET (inbuffer) = (j - 1) * 2 * samples;
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GST_BUFFER_OFFSET_END (inbuffer) = j * 2 * samples + samples;
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gst_buffer_set_caps (inbuffer, caps);
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p = (gint16 *) GST_BUFFER_DATA (inbuffer);
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/* create a 16 bit signed ramp */
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for (i = 0; i < samples; ++i) {
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*p = -32767 + i * (65535 / samples);
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++p;
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*p = -32767 + i * (65535 / samples);
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++p;
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}
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GST_DEBUG ("Sending Buffer time:%" G_GUINT64_FORMAT " duration:%"
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G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
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G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (inbuffer),
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GST_BUFFER_DURATION (inbuffer), GST_BUFFER_IS_DISCONT (inbuffer),
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GST_BUFFER_OFFSET (inbuffer), GST_BUFFER_OFFSET_END (inbuffer));
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/* pushing gives away my reference ... */
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fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
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/* check if the timestamp of the pushed buffer matches the incoming one */
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outbuffer = g_list_nth_data (buffers, g_list_length (buffers) - 1);
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fail_if (outbuffer == NULL);
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fail_unless_equals_uint64 (ints, GST_BUFFER_TIMESTAMP (outbuffer));
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GST_DEBUG ("Got Buffer time:%" G_GUINT64_FORMAT " duration:%"
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G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
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G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (outbuffer),
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GST_BUFFER_DURATION (outbuffer), GST_BUFFER_IS_DISCONT (outbuffer),
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GST_BUFFER_OFFSET (outbuffer), GST_BUFFER_OFFSET_END (outbuffer));
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if (j > 1) {
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fail_unless (GST_BUFFER_IS_DISCONT (outbuffer),
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"expected discont for buffer #%d", j);
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}
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}
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/* cleanup */
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gst_caps_unref (caps);
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cleanup_speexresample (speexresample);
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}
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GST_START_TEST (test_discont_stream)
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{
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/* integral scalings */
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test_discont_stream_instance (48000, 24000, 500, 20);
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test_discont_stream_instance (48000, 12000, 500, 20);
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test_discont_stream_instance (12000, 24000, 500, 20);
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test_discont_stream_instance (12000, 48000, 500, 20);
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/* non-integral scalings */
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test_discont_stream_instance (44100, 8000, 500, 20);
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test_discont_stream_instance (8000, 44100, 500, 20);
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/* wacky scalings */
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test_discont_stream_instance (12345, 54321, 500, 20);
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test_discont_stream_instance (101, 99, 500, 20);
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}
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GST_END_TEST;
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GST_START_TEST (test_reuse)
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{
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GstElement *speexresample;
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GstEvent *newseg;
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GstBuffer *inbuffer;
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GstCaps *caps;
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speexresample = setup_speexresample (1, 9343, 48000, 16, FALSE);
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caps = gst_pad_get_negotiated_caps (mysrcpad);
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fail_unless (gst_caps_is_fixed (caps));
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fail_unless (gst_element_set_state (speexresample,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
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"could not set to playing");
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newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
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fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
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inbuffer = gst_buffer_new_and_alloc (9343 * 4);
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memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
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GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
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GST_BUFFER_TIMESTAMP (inbuffer) = 0;
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GST_BUFFER_OFFSET (inbuffer) = 0;
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gst_buffer_set_caps (inbuffer, caps);
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/* pushing gives away my reference ... */
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fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
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/* ... but it ends up being collected on the global buffer list */
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fail_unless_equals_int (g_list_length (buffers), 1);
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/* now reset and try again ... */
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fail_unless (gst_element_set_state (speexresample,
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GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
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fail_unless (gst_element_set_state (speexresample,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
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"could not set to playing");
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newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
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fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
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inbuffer = gst_buffer_new_and_alloc (9343 * 4);
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memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
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GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
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GST_BUFFER_TIMESTAMP (inbuffer) = 0;
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GST_BUFFER_OFFSET (inbuffer) = 0;
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gst_buffer_set_caps (inbuffer, caps);
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fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
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/* ... it also ends up being collected on the global buffer list. If we
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* now have more than 2 buffers, then speexresample probably didn't clean
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* up its internal buffer properly and tried to push the remaining samples
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* when it got the second NEWSEGMENT event */
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fail_unless_equals_int (g_list_length (buffers), 2);
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cleanup_speexresample (speexresample);
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gst_caps_unref (caps);
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}
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GST_END_TEST;
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GST_START_TEST (test_shutdown)
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{
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GstElement *pipeline, *src, *cf1, *ar, *cf2, *sink;
|
|
GstCaps *caps;
|
|
guint i;
|
|
|
|
/* create pipeline, force speexresample to actually resample */
|
|
pipeline = gst_pipeline_new (NULL);
|
|
|
|
src = gst_check_setup_element ("audiotestsrc");
|
|
cf1 = gst_check_setup_element ("capsfilter");
|
|
ar = gst_check_setup_element ("audioresample");
|
|
cf2 = gst_check_setup_element ("capsfilter");
|
|
g_object_set (cf2, "name", "capsfilter2", NULL);
|
|
sink = gst_check_setup_element ("fakesink");
|
|
|
|
caps =
|
|
gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT, 11025, NULL);
|
|
g_object_set (cf1, "caps", caps, NULL);
|
|
gst_caps_unref (caps);
|
|
|
|
caps =
|
|
gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT, 48000, NULL);
|
|
g_object_set (cf2, "caps", caps, NULL);
|
|
gst_caps_unref (caps);
|
|
|
|
/* don't want to sync against the clock, the more throughput the better */
|
|
g_object_set (src, "is-live", FALSE, NULL);
|
|
g_object_set (sink, "sync", FALSE, NULL);
|
|
|
|
gst_bin_add_many (GST_BIN (pipeline), src, cf1, ar, cf2, sink, NULL);
|
|
fail_if (!gst_element_link_many (src, cf1, ar, cf2, sink, NULL));
|
|
|
|
/* now, wait until pipeline is running and then shut it down again; repeat */
|
|
for (i = 0; i < 20; ++i) {
|
|
gst_element_set_state (pipeline, GST_STATE_PAUSED);
|
|
gst_element_get_state (pipeline, NULL, NULL, -1);
|
|
gst_element_set_state (pipeline, GST_STATE_PLAYING);
|
|
g_usleep (100);
|
|
gst_element_set_state (pipeline, GST_STATE_NULL);
|
|
}
|
|
|
|
gst_object_unref (pipeline);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static GstFlowReturn
|
|
live_switch_alloc_only_48000 (GstPad * pad, guint64 offset,
|
|
guint size, GstCaps * caps, GstBuffer ** buf)
|
|
{
|
|
GstStructure *structure;
|
|
gint rate;
|
|
gint channels;
|
|
GstCaps *desired;
|
|
|
|
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));
|
|
|
|
if (rate < 48000)
|
|
return GST_FLOW_NOT_NEGOTIATED;
|
|
|
|
desired = gst_caps_copy (caps);
|
|
gst_caps_set_simple (desired, "rate", G_TYPE_INT, 48000, NULL);
|
|
|
|
*buf = gst_buffer_new_and_alloc (channels * 48000);
|
|
gst_buffer_set_caps (*buf, desired);
|
|
gst_caps_unref (desired);
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static GstCaps *
|
|
live_switch_get_sink_caps (GstPad * pad)
|
|
{
|
|
GstCaps *result;
|
|
|
|
result = gst_caps_copy (GST_PAD_CAPS (pad));
|
|
|
|
gst_caps_set_simple (result,
|
|
"rate", GST_TYPE_INT_RANGE, 48000, G_MAXINT, NULL);
|
|
|
|
return result;
|
|
}
|
|
|
|
static void
|
|
live_switch_push (int rate, GstCaps * caps)
|
|
{
|
|
GstBuffer *inbuffer;
|
|
GstCaps *desired;
|
|
GList *l;
|
|
|
|
desired = gst_caps_copy (caps);
|
|
gst_caps_set_simple (desired, "rate", G_TYPE_INT, rate, NULL);
|
|
gst_pad_set_caps (mysrcpad, desired);
|
|
|
|
fail_unless (gst_pad_alloc_buffer_and_set_caps (mysrcpad,
|
|
GST_BUFFER_OFFSET_NONE, rate * 4, desired, &inbuffer) == GST_FLOW_OK);
|
|
|
|
/* When the basetransform hits the non-configured case it always
|
|
* returns a buffer with exactly the same caps as we requested so the actual
|
|
* renegotiation (if needed) will be done in the _chain*/
|
|
fail_unless (inbuffer != NULL);
|
|
GST_DEBUG ("desired: %" GST_PTR_FORMAT ".... got: %" GST_PTR_FORMAT,
|
|
desired, GST_BUFFER_CAPS (inbuffer));
|
|
fail_unless (gst_caps_is_equal (desired, GST_BUFFER_CAPS (inbuffer)));
|
|
|
|
memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
|
|
GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
|
|
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
|
|
GST_BUFFER_OFFSET (inbuffer) = 0;
|
|
|
|
/* pushing gives away my reference ... */
|
|
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
|
|
|
|
/* ... but it ends up being collected on the global buffer list */
|
|
fail_unless_equals_int (g_list_length (buffers), 1);
|
|
|
|
for (l = buffers; l; l = l->next) {
|
|
GstBuffer *buffer = GST_BUFFER (l->data);
|
|
|
|
gst_buffer_unref (buffer);
|
|
}
|
|
|
|
g_list_free (buffers);
|
|
buffers = NULL;
|
|
|
|
gst_caps_unref (desired);
|
|
}
|
|
|
|
GST_START_TEST (test_live_switch)
|
|
{
|
|
GstElement *speexresample;
|
|
GstEvent *newseg;
|
|
GstCaps *caps;
|
|
|
|
speexresample = setup_speexresample (4, 48000, 48000, 16, FALSE);
|
|
|
|
/* Let the sinkpad act like something that can only handle things of
|
|
* rate 48000- and can only allocate buffers for that rate, but if someone
|
|
* tries to get a buffer with a rate higher then 48000 tries to renegotiate
|
|
* */
|
|
gst_pad_set_bufferalloc_function (mysinkpad, live_switch_alloc_only_48000);
|
|
gst_pad_set_getcaps_function (mysinkpad, live_switch_get_sink_caps);
|
|
|
|
gst_pad_use_fixed_caps (mysrcpad);
|
|
|
|
caps = gst_pad_get_negotiated_caps (mysrcpad);
|
|
fail_unless (gst_caps_is_fixed (caps));
|
|
|
|
fail_unless (gst_element_set_state (speexresample,
|
|
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
|
|
"could not set to playing");
|
|
|
|
newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
|
|
fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
|
|
|
|
/* downstream can provide the requested rate, a buffer alloc will be passed
|
|
* on */
|
|
live_switch_push (48000, caps);
|
|
|
|
/* Downstream can never accept this rate, buffer alloc isn't passed on */
|
|
live_switch_push (40000, caps);
|
|
|
|
/* Downstream can provide the requested rate but will re-negotiate */
|
|
live_switch_push (50000, caps);
|
|
|
|
cleanup_speexresample (speexresample);
|
|
gst_caps_unref (caps);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
#ifndef GST_DISABLE_PARSE
|
|
|
|
static GMainLoop *loop;
|
|
static gint messages = 0;
|
|
|
|
static void
|
|
element_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
|
|
{
|
|
gchar *s;
|
|
|
|
s = gst_structure_to_string (gst_message_get_structure (message));
|
|
GST_DEBUG ("Received message: %s", s);
|
|
g_free (s);
|
|
|
|
messages++;
|
|
}
|
|
|
|
static void
|
|
eos_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
|
|
{
|
|
GST_DEBUG ("Received eos");
|
|
g_main_loop_quit (loop);
|
|
}
|
|
|
|
static void
|
|
test_pipeline (gint width, gboolean fp, gint inrate, gint outrate, gint quality)
|
|
{
|
|
GstElement *pipeline;
|
|
GstBus *bus;
|
|
GError *error = NULL;
|
|
gchar *pipe_str;
|
|
|
|
pipe_str =
|
|
g_strdup_printf
|
|
("audiotestsrc num-buffers=10 ! audioconvert ! audio/x-raw-%s,rate=%d,width=%d,channels=2 ! audioresample quality=%d ! audio/x-raw-%s,rate=%d,width=%d ! identity check-imperfect-timestamp=TRUE ! fakesink",
|
|
(fp) ? "float" : "int", inrate, width, quality, (fp) ? "float" : "int",
|
|
outrate, width);
|
|
|
|
pipeline = gst_parse_launch (pipe_str, &error);
|
|
fail_unless (pipeline != NULL, "Error parsing pipeline: %s",
|
|
error ? error->message : "(invalid error)");
|
|
g_free (pipe_str);
|
|
|
|
bus = gst_element_get_bus (pipeline);
|
|
fail_if (bus == NULL);
|
|
gst_bus_add_signal_watch (bus);
|
|
g_signal_connect (bus, "message::element", (GCallback) element_message_cb,
|
|
NULL);
|
|
g_signal_connect (bus, "message::eos", (GCallback) eos_message_cb, NULL);
|
|
|
|
gst_element_set_state (pipeline, GST_STATE_PLAYING);
|
|
|
|
/* run until we receive EOS */
|
|
loop = g_main_loop_new (NULL, FALSE);
|
|
|
|
g_main_loop_run (loop);
|
|
|
|
g_main_loop_unref (loop);
|
|
loop = NULL;
|
|
|
|
gst_element_set_state (pipeline, GST_STATE_NULL);
|
|
|
|
fail_if (messages > 0, "Received imperfect timestamp messages");
|
|
gst_object_unref (pipeline);
|
|
}
|
|
|
|
GST_START_TEST (test_pipelines)
|
|
{
|
|
gint quality;
|
|
|
|
/* Test qualities 0, 5 and 10 */
|
|
for (quality = 0; quality < 11; quality += 5) {
|
|
test_pipeline (8, FALSE, 44100, 48000, quality);
|
|
test_pipeline (8, FALSE, 48000, 44100, quality);
|
|
|
|
test_pipeline (16, FALSE, 44100, 48000, quality);
|
|
test_pipeline (16, FALSE, 48000, 44100, quality);
|
|
|
|
test_pipeline (24, FALSE, 44100, 48000, quality);
|
|
test_pipeline (24, FALSE, 48000, 44100, quality);
|
|
|
|
test_pipeline (32, FALSE, 44100, 48000, quality);
|
|
test_pipeline (32, FALSE, 48000, 44100, quality);
|
|
|
|
test_pipeline (32, TRUE, 44100, 48000, quality);
|
|
test_pipeline (32, TRUE, 48000, 44100, quality);
|
|
|
|
test_pipeline (64, TRUE, 44100, 48000, quality);
|
|
test_pipeline (64, TRUE, 48000, 44100, quality);
|
|
}
|
|
}
|
|
|
|
GST_END_TEST;
|
|
#endif
|
|
|
|
static Suite *
|
|
speexresample_suite (void)
|
|
{
|
|
Suite *s = suite_create ("speexresample");
|
|
TCase *tc_chain = tcase_create ("general");
|
|
|
|
suite_add_tcase (s, tc_chain);
|
|
tcase_add_test (tc_chain, test_perfect_stream);
|
|
tcase_add_test (tc_chain, test_discont_stream);
|
|
tcase_add_test (tc_chain, test_reuse);
|
|
tcase_add_test (tc_chain, test_shutdown);
|
|
tcase_add_test (tc_chain, test_live_switch);
|
|
|
|
#ifndef GST_DISABLE_PARSE
|
|
tcase_set_timeout (tc_chain, 360);
|
|
tcase_add_test (tc_chain, test_pipelines);
|
|
#endif
|
|
|
|
return s;
|
|
}
|
|
|
|
GST_CHECK_MAIN (speexresample);
|