--- /dev/null
+/* GStreamer
+ *
+ * unit test for audioresample, based on the audioresample unit test
+ *
+ * Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
+ * Copyright (C) <2006> Tim-Philipp Müller <tim at centricular net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#include <unistd.h>
+
+#include <gst/check/gstcheck.h>
+
+#include <gst/audio/audio.h>
+
+/* For ease of programming we use globals to keep refs for our floating
+ * src and sink pads we create; otherwise we always have to do get_pad,
+ * get_peer, and then remove references in every test function */
+static GstPad *mysrcpad, *mysinkpad;
+
+#define RESAMPLE_CAPS_FLOAT \
+ "audio/x-raw-float, " \
+ "channels = (int) [ 1, MAX ], " \
+ "rate = (int) [ 1, MAX ], " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) { 32, 64 }"
+
+#define RESAMPLE_CAPS_INT \
+ "audio/x-raw-int, " \
+ "channels = (int) [ 1, MAX ], " \
+ "rate = (int) [ 1, MAX ], " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 16, " \
+ "depth = (int) 16, " \
+ "signed = (bool) TRUE"
+
+#define RESAMPLE_CAPS_TEMPLATE_STRING \
+ RESAMPLE_CAPS_FLOAT " ; " \
+ RESAMPLE_CAPS_INT
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (RESAMPLE_CAPS_TEMPLATE_STRING)
+ );
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (RESAMPLE_CAPS_TEMPLATE_STRING)
+ );
+
+static GstElement *
+setup_audioresample (int channels, int inrate, int outrate, int width,
+ gboolean fp)
+{
+ GstElement *audioresample;
+ GstCaps *caps;
+ GstStructure *structure;
+
+ GST_DEBUG ("setup_audioresample");
+ audioresample = gst_check_setup_element ("audioresample");
+
+ if (fp)
+ caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
+ else
+ caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
+ structure = gst_caps_get_structure (caps, 0);
+ gst_structure_set (structure, "channels", G_TYPE_INT, channels,
+ "rate", G_TYPE_INT, inrate, "width", G_TYPE_INT, width, NULL);
+ if (!fp)
+ gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
+ fail_unless (gst_caps_is_fixed (caps));
+
+ fail_unless (gst_element_set_state (audioresample,
+ GST_STATE_PAUSED) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to paused");
+
+ mysrcpad = gst_check_setup_src_pad (audioresample, &srctemplate, caps);
+ gst_pad_set_caps (mysrcpad, caps);
+ gst_caps_unref (caps);
+
+ if (fp)
+ caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
+ else
+ caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
+ structure = gst_caps_get_structure (caps, 0);
+ gst_structure_set (structure, "channels", G_TYPE_INT, channels,
+ "rate", G_TYPE_INT, outrate, "width", G_TYPE_INT, width, NULL);
+ if (!fp)
+ gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
+ fail_unless (gst_caps_is_fixed (caps));
+
+ mysinkpad = gst_check_setup_sink_pad (audioresample, &sinktemplate, caps);
+ /* this installs a getcaps func that will always return the caps we set
+ * later */
+ gst_pad_set_caps (mysinkpad, caps);
+ gst_pad_use_fixed_caps (mysinkpad);
+
+ gst_pad_set_active (mysinkpad, TRUE);
+ gst_pad_set_active (mysrcpad, TRUE);
+
+ gst_caps_unref (caps);
+
+ return audioresample;
+}
+
+static void
+cleanup_audioresample (GstElement * audioresample)
+{
+ GST_DEBUG ("cleanup_audioresample");
+
+ fail_unless (gst_element_set_state (audioresample,
+ GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
+
+ gst_pad_set_active (mysrcpad, FALSE);
+ gst_pad_set_active (mysinkpad, FALSE);
+ gst_check_teardown_src_pad (audioresample);
+ gst_check_teardown_sink_pad (audioresample);
+ gst_check_teardown_element (audioresample);
+ gst_check_drop_buffers ();
+}
+
+static void
+fail_unless_perfect_stream (void)
+{
+ guint64 timestamp = 0L, duration = 0L;
+ guint64 offset = 0L, offset_end = 0L;
+
+ GList *l;
+ GstBuffer *buffer;
+
+ for (l = buffers; l; l = l->next) {
+ buffer = GST_BUFFER (l->data);
+ ASSERT_BUFFER_REFCOUNT (buffer, "buffer", 1);
+ GST_DEBUG ("buffer timestamp %" G_GUINT64_FORMAT ", duration %"
+ G_GUINT64_FORMAT " offset %" G_GUINT64_FORMAT " offset_end %"
+ G_GUINT64_FORMAT,
+ GST_BUFFER_TIMESTAMP (buffer),
+ GST_BUFFER_DURATION (buffer),
+ GST_BUFFER_OFFSET (buffer), GST_BUFFER_OFFSET_END (buffer));
+
+ fail_unless_equals_uint64 (timestamp, GST_BUFFER_TIMESTAMP (buffer));
+ fail_unless_equals_uint64 (offset, GST_BUFFER_OFFSET (buffer));
+ duration = GST_BUFFER_DURATION (buffer);
+ offset_end = GST_BUFFER_OFFSET_END (buffer);
+
+ timestamp += duration;
+ offset = offset_end;
+ gst_buffer_unref (buffer);
+ }
+ g_list_free (buffers);
+ buffers = NULL;
+}
+
+/* this tests that the output is a perfect stream if the input is */
+static void
+test_perfect_stream_instance (int inrate, int outrate, int samples,
+ int numbuffers)
+{
+ GstElement *audioresample;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ guint64 offset = 0;
+
+ int i, j;
+ gint16 *p;
+
+ audioresample = setup_audioresample (2, inrate, outrate, 16, FALSE);
+ caps = gst_pad_get_negotiated_caps (mysrcpad);
+ fail_unless (gst_caps_is_fixed (caps));
+
+ fail_unless (gst_element_set_state (audioresample,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ for (j = 1; j <= numbuffers; ++j) {
+
+ inbuffer = gst_buffer_new_and_alloc (samples * 4);
+ GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (samples, inrate);
+ GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1);
+ GST_BUFFER_OFFSET (inbuffer) = offset;
+ offset += samples;
+ GST_BUFFER_OFFSET_END (inbuffer) = offset;
+
+ gst_buffer_set_caps (inbuffer, caps);
+
+ p = (gint16 *) GST_BUFFER_DATA (inbuffer);
+
+ /* create a 16 bit signed ramp */
+ for (i = 0; i < samples; ++i) {
+ *p = -32767 + i * (65535 / samples);
+ ++p;
+ *p = -32767 + i * (65535 / samples);
+ ++p;
+ }
+
+ /* 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), j);
+ }
+
+ /* FIXME: we should make audioresample handle eos by flushing out the last
+ * samples, which will give us one more, small, buffer */
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+ ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
+
+ fail_unless_perfect_stream ();
+
+ /* cleanup */
+ gst_caps_unref (caps);
+ cleanup_audioresample (audioresample);
+}
+
+
+/* make sure that outgoing buffers are contiguous in timestamp/duration and
+ * offset/offsetend
+ */
+GST_START_TEST (test_perfect_stream)
+{
+ /* integral scalings */
+ test_perfect_stream_instance (48000, 24000, 500, 20);
+ test_perfect_stream_instance (48000, 12000, 500, 20);
+ test_perfect_stream_instance (12000, 24000, 500, 20);
+ test_perfect_stream_instance (12000, 48000, 500, 20);
+
+ /* non-integral scalings */
+ test_perfect_stream_instance (44100, 8000, 500, 20);
+ test_perfect_stream_instance (8000, 44100, 500, 20);
+
+ /* wacky scalings */
+ test_perfect_stream_instance (12345, 54321, 500, 20);
+ test_perfect_stream_instance (101, 99, 500, 20);
+}
+
+GST_END_TEST;
+
+/* this tests that the output is a correct discontinuous stream
+ * if the input is; ie input drops in time come out the same way */
+static void
+test_discont_stream_instance (int inrate, int outrate, int samples,
+ int numbuffers)
+{
+ GstElement *audioresample;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ GstClockTime ints;
+
+ int i, j;
+ gint16 *p;
+
+ GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d",
+ inrate, outrate, samples, numbuffers);
+
+ audioresample = setup_audioresample (2, inrate, outrate, 16, FALSE);
+ caps = gst_pad_get_negotiated_caps (mysrcpad);
+ fail_unless (gst_caps_is_fixed (caps));
+
+ fail_unless (gst_element_set_state (audioresample,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ for (j = 1; j <= numbuffers; ++j) {
+
+ inbuffer = gst_buffer_new_and_alloc (samples * 4);
+ GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate;
+ /* "drop" half the buffers */
+ ints = GST_BUFFER_DURATION (inbuffer) * 2 * (j - 1);
+ GST_BUFFER_TIMESTAMP (inbuffer) = ints;
+ GST_BUFFER_OFFSET (inbuffer) = (j - 1) * 2 * samples;
+ GST_BUFFER_OFFSET_END (inbuffer) = j * 2 * samples + samples;
+
+ gst_buffer_set_caps (inbuffer, caps);
+
+ p = (gint16 *) GST_BUFFER_DATA (inbuffer);
+
+ /* create a 16 bit signed ramp */
+ for (i = 0; i < samples; ++i) {
+ *p = -32767 + i * (65535 / samples);
+ ++p;
+ *p = -32767 + i * (65535 / samples);
+ ++p;
+ }
+
+ GST_DEBUG ("Sending Buffer time:%" G_GUINT64_FORMAT " duration:%"
+ G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
+ G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (inbuffer),
+ GST_BUFFER_DURATION (inbuffer), GST_BUFFER_IS_DISCONT (inbuffer),
+ GST_BUFFER_OFFSET (inbuffer), GST_BUFFER_OFFSET_END (inbuffer));
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+
+ /* check if the timestamp of the pushed buffer matches the incoming one */
+ outbuffer = g_list_nth_data (buffers, g_list_length (buffers) - 1);
+ fail_if (outbuffer == NULL);
+ fail_unless_equals_uint64 (ints, GST_BUFFER_TIMESTAMP (outbuffer));
+ GST_DEBUG ("Got Buffer time:%" G_GUINT64_FORMAT " duration:%"
+ G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
+ G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (outbuffer),
+ GST_BUFFER_DURATION (outbuffer), GST_BUFFER_IS_DISCONT (outbuffer),
+ GST_BUFFER_OFFSET (outbuffer), GST_BUFFER_OFFSET_END (outbuffer));
+ if (j > 1) {
+ fail_unless (GST_BUFFER_IS_DISCONT (outbuffer),
+ "expected discont for buffer #%d", j);
+ }
+ }
+
+ /* cleanup */
+ gst_caps_unref (caps);
+ cleanup_audioresample (audioresample);
+}
+
+GST_START_TEST (test_discont_stream)
+{
+ /* integral scalings */
+ test_discont_stream_instance (48000, 24000, 5000, 20);
+ test_discont_stream_instance (48000, 12000, 5000, 20);
+ test_discont_stream_instance (12000, 24000, 5000, 20);
+ test_discont_stream_instance (12000, 48000, 5000, 20);
+
+ /* non-integral scalings */
+ test_discont_stream_instance (44100, 8000, 5000, 20);
+ test_discont_stream_instance (8000, 44100, 5000, 20);
+
+ /* wacky scalings */
+ test_discont_stream_instance (12345, 54321, 5000, 20);
+ test_discont_stream_instance (101, 99, 5000, 20);
+}
+
+GST_END_TEST;
+
+
+
+GST_START_TEST (test_reuse)
+{
+ GstElement *audioresample;
+ GstEvent *newseg;
+ GstBuffer *inbuffer;
+ GstCaps *caps;
+
+ audioresample = setup_audioresample (1, 9343, 48000, 16, FALSE);
+ caps = gst_pad_get_negotiated_caps (mysrcpad);
+ fail_unless (gst_caps_is_fixed (caps));
+
+ fail_unless (gst_element_set_state (audioresample,
+ 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);
+
+ inbuffer = gst_buffer_new_and_alloc (9343 * 4);
+ 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;
+ gst_buffer_set_caps (inbuffer, caps);
+
+ /* 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);
+
+ /* now reset and try again ... */
+ fail_unless (gst_element_set_state (audioresample,
+ GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
+
+ fail_unless (gst_element_set_state (audioresample,
+ 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);
+
+ inbuffer = gst_buffer_new_and_alloc (9343 * 4);
+ 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;
+ gst_buffer_set_caps (inbuffer, caps);
+
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+
+ /* ... it also ends up being collected on the global buffer list. If we
+ * now have more than 2 buffers, then audioresample probably didn't clean
+ * up its internal buffer properly and tried to push the remaining samples
+ * when it got the second NEWSEGMENT event */
+ fail_unless_equals_int (g_list_length (buffers), 2);
+
+ cleanup_audioresample (audioresample);
+ gst_caps_unref (caps);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_shutdown)
+{
+ GstElement *pipeline, *src, *cf1, *ar, *cf2, *sink;
+ GstCaps *caps;
+ guint i;
+
+ /* create pipeline, force audioresample 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 *audioresample;
+ GstEvent *newseg;
+ GstCaps *caps;
+
+ audioresample = setup_audioresample (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 (audioresample,
+ 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_audioresample (audioresample);
+ 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) {
+ GST_DEBUG ("Checking with quality %d", quality);
+
+ 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;
+
+GST_START_TEST (test_preference_passthrough)
+{
+ GstStateChangeReturn ret;
+ GstElement *pipeline, *src;
+ GstStructure *s;
+ GstMessage *msg;
+ GstCaps *caps;
+ GstPad *pad;
+ GstBus *bus;
+ GError *error = NULL;
+ gint rate = 0;
+
+ pipeline = gst_parse_launch ("audiotestsrc num-buffers=1 name=src ! "
+ "audioresample ! audio/x-raw-int,channels=1,width=16,depth=16,"
+ "endianness=BYTE_ORDER,signed=true,rate=8000 ! "
+ "fakesink can-activate-pull=false", &error);
+ fail_unless (pipeline != NULL, "Error parsing pipeline: %s",
+ error ? error->message : "(invalid error)");
+
+ ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
+ fail_unless_equals_int (ret, GST_STATE_CHANGE_ASYNC);
+
+ /* run until we receive EOS */
+ bus = gst_element_get_bus (pipeline);
+ fail_if (bus == NULL);
+ msg = gst_bus_timed_pop_filtered (bus, -1, GST_MESSAGE_EOS);
+ gst_message_unref (msg);
+ gst_object_unref (bus);
+
+ src = gst_bin_get_by_name (GST_BIN (pipeline), "src");
+ fail_unless (src != NULL);
+ pad = gst_element_get_static_pad (src, "src");
+ fail_unless (pad != NULL);
+ caps = gst_pad_get_negotiated_caps (pad);
+ GST_LOG ("negotiated audiotestsrc caps: %" GST_PTR_FORMAT, caps);
+ fail_unless (caps != NULL);
+ s = gst_caps_get_structure (caps, 0);
+ fail_unless (gst_structure_get_int (s, "rate", &rate));
+ /* there's no need to resample, audiotestsrc supports any rate, so make
+ * sure audioresample provided upstream with the right caps to negotiate
+ * this correctly */
+ fail_unless_equals_int (rate, 8000);
+ gst_caps_unref (caps);
+ gst_object_unref (pad);
+ gst_object_unref (src);
+
+ gst_element_set_state (pipeline, GST_STATE_NULL);
+ gst_object_unref (pipeline);
+}
+
+GST_END_TEST;
+
+#endif
+
+static void
+_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
+{
+ GMainLoop *loop = user_data;
+
+ switch (GST_MESSAGE_TYPE (message)) {
+ case GST_MESSAGE_ERROR:
+ case GST_MESSAGE_WARNING:
+ g_assert_not_reached ();
+ break;
+ case GST_MESSAGE_EOS:
+ g_main_loop_quit (loop);
+ break;
+ default:
+ break;
+ }
+}
+
+typedef struct
+{
+ guint64 latency;
+ GstClockTime in_ts;
+
+ GstClockTime next_out_ts;
+ guint64 next_out_off;
+
+ guint64 in_buffer_count, out_buffer_count;
+} TimestampDriftCtx;
+
+static void
+fakesink_handoff_cb (GstElement * object, GstBuffer * buffer, GstPad * pad,
+ gpointer user_data)
+{
+ TimestampDriftCtx *ctx = user_data;
+
+ ctx->out_buffer_count++;
+ if (ctx->latency == GST_CLOCK_TIME_NONE) {
+ ctx->latency = 1000 - GST_BUFFER_SIZE (buffer) / 8;
+ }
+
+ /* Check if we have a perfectly timestampped stream */
+ if (ctx->next_out_ts != GST_CLOCK_TIME_NONE)
+ fail_unless (ctx->next_out_ts == GST_BUFFER_TIMESTAMP (buffer),
+ "expected timestamp %" GST_TIME_FORMAT " got timestamp %"
+ GST_TIME_FORMAT, GST_TIME_ARGS (ctx->next_out_ts),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
+
+ /* Check if we have a perfectly offsetted stream */
+ fail_unless (GST_BUFFER_OFFSET_END (buffer) ==
+ GST_BUFFER_OFFSET (buffer) + GST_BUFFER_SIZE (buffer) / 8,
+ "expected offset end %" G_GUINT64_FORMAT " got offset end %"
+ G_GUINT64_FORMAT,
+ GST_BUFFER_OFFSET (buffer) + GST_BUFFER_SIZE (buffer) / 8,
+ GST_BUFFER_OFFSET_END (buffer));
+ if (ctx->next_out_off != GST_BUFFER_OFFSET_NONE) {
+ fail_unless (GST_BUFFER_OFFSET (buffer) == ctx->next_out_off,
+ "expected offset %" G_GUINT64_FORMAT " got offset %" G_GUINT64_FORMAT,
+ ctx->next_out_off, GST_BUFFER_OFFSET (buffer));
+ }
+
+ if (ctx->in_buffer_count != ctx->out_buffer_count) {
+ GST_INFO ("timestamp %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
+ }
+
+ if (ctx->in_ts != GST_CLOCK_TIME_NONE && ctx->in_buffer_count > 1
+ && ctx->in_buffer_count == ctx->out_buffer_count) {
+ fail_unless (GST_BUFFER_TIMESTAMP (buffer) ==
+ ctx->in_ts - gst_util_uint64_scale_round (ctx->latency, GST_SECOND,
+ 4096),
+ "expected output timestamp %" GST_TIME_FORMAT " (%" G_GUINT64_FORMAT
+ ") got output timestamp %" GST_TIME_FORMAT " (%" G_GUINT64_FORMAT ")",
+ GST_TIME_ARGS (ctx->in_ts - gst_util_uint64_scale_round (ctx->latency,
+ GST_SECOND, 4096)),
+ ctx->in_ts - gst_util_uint64_scale_round (ctx->latency, GST_SECOND,
+ 4096), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
+ GST_BUFFER_TIMESTAMP (buffer));
+ }
+
+ ctx->next_out_ts =
+ GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer);
+ ctx->next_out_off = GST_BUFFER_OFFSET_END (buffer);
+}
+
+static void
+identity_handoff_cb (GstElement * object, GstBuffer * buffer,
+ gpointer user_data)
+{
+ TimestampDriftCtx *ctx = user_data;
+
+ ctx->in_ts = GST_BUFFER_TIMESTAMP (buffer);
+ ctx->in_buffer_count++;
+}
+
+GST_START_TEST (test_timestamp_drift)
+{
+ TimestampDriftCtx ctx =
+ { GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE,
+ GST_BUFFER_OFFSET_NONE, 0, 0
+ };
+ GstElement *pipeline;
+ GstElement *audiotestsrc, *capsfilter1, *identity, *audioresample,
+ *capsfilter2, *fakesink;
+ GstBus *bus;
+ GMainLoop *loop;
+ GstCaps *caps;
+
+ pipeline = gst_pipeline_new ("pipeline");
+ fail_unless (pipeline != NULL);
+
+ audiotestsrc = gst_element_factory_make ("audiotestsrc", "src");
+ fail_unless (audiotestsrc != NULL);
+ g_object_set (G_OBJECT (audiotestsrc), "num-buffers", 10000,
+ "samplesperbuffer", 4000, NULL);
+
+ capsfilter1 = gst_element_factory_make ("capsfilter", "capsfilter1");
+ fail_unless (capsfilter1 != NULL);
+ caps =
+ gst_caps_from_string
+ ("audio/x-raw-float, channels=1, width=64, rate=16384");
+ g_object_set (G_OBJECT (capsfilter1), "caps", caps, NULL);
+ gst_caps_unref (caps);
+
+ identity = gst_element_factory_make ("identity", "identity");
+ fail_unless (identity != NULL);
+ g_object_set (G_OBJECT (identity), "sync", FALSE, "signal-handoffs", TRUE,
+ NULL);
+ g_signal_connect (identity, "handoff", (GCallback) identity_handoff_cb, &ctx);
+
+ audioresample = gst_element_factory_make ("audioresample", "resample");
+ fail_unless (audioresample != NULL);
+ capsfilter2 = gst_element_factory_make ("capsfilter", "capsfilter2");
+ fail_unless (capsfilter2 != NULL);
+ caps =
+ gst_caps_from_string
+ ("audio/x-raw-float, channels=1, width=64, rate=4096");
+ g_object_set (G_OBJECT (capsfilter2), "caps", caps, NULL);
+ gst_caps_unref (caps);
+
+ fakesink = gst_element_factory_make ("fakesink", "sink");
+ fail_unless (fakesink != NULL);
+ g_object_set (G_OBJECT (fakesink), "sync", FALSE, "async", FALSE,
+ "signal-handoffs", TRUE, NULL);
+ g_signal_connect (fakesink, "handoff", (GCallback) fakesink_handoff_cb, &ctx);
+
+
+ gst_bin_add_many (GST_BIN (pipeline), audiotestsrc, capsfilter1, identity,
+ audioresample, capsfilter2, fakesink, NULL);
+ fail_unless (gst_element_link_many (audiotestsrc, capsfilter1, identity,
+ audioresample, capsfilter2, fakesink, NULL));
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ bus = gst_element_get_bus (pipeline);
+ gst_bus_add_signal_watch (bus);
+ g_signal_connect (bus, "message", (GCallback) _message_cb, loop);
+
+ fail_unless (gst_element_set_state (pipeline,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
+ g_main_loop_run (loop);
+
+ fail_unless (gst_element_set_state (pipeline,
+ GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS);
+ g_main_loop_unref (loop);
+ gst_object_unref (pipeline);
+
+} GST_END_TEST;
+
+static Suite *
+audioresample_suite (void)
+{
+ Suite *s = suite_create ("audioresample");
+ 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);
+ tcase_add_test (tc_chain, test_timestamp_drift);
+
+#ifndef GST_DISABLE_PARSE
+ tcase_set_timeout (tc_chain, 360);
+ tcase_add_test (tc_chain, test_pipelines);
+ tcase_add_test (tc_chain, test_preference_passthrough);
+#endif
+
+ return s;
+}
+
+GST_CHECK_MAIN (audioresample);
projects / mafwsubrenderer / blobdiff