| /* 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., 51 Franklin St, Fifth Floor, |
| * Boston, MA 02110-1301, USA. |
| */ |
| |
| #include <unistd.h> |
| |
| #include <gst/check/gstcheck.h> |
| |
| #include <gst/audio/audio.h> |
| |
| #include <gst/fft/gstfft.h> |
| #include <gst/fft/gstffts16.h> |
| #include <gst/fft/gstffts32.h> |
| #include <gst/fft/gstfftf32.h> |
| #include <gst/fft/gstfftf64.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; |
| |
| #if G_BYTE_ORDER == G_LITTLE_ENDIAN |
| #define FORMATS "{ F32LE, F64LE, S16LE, S32LE }" |
| #else |
| #define FORMATS "{ F32BE, F64BE, S16BE, S32BE }" |
| #endif |
| |
| #define RESAMPLE_CAPS \ |
| "audio/x-raw, " \ |
| "format = (string) "FORMATS", " \ |
| "channels = (int) [ 1, MAX ], " \ |
| "rate = (int) [ 1, MAX ], " \ |
| "layout = (string) interleaved" |
| |
| static GstElement * |
| setup_audioresample (int channels, guint64 mask, int inrate, int outrate, |
| const gchar * format) |
| { |
| GstPadTemplate *sinktemplate; |
| static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", |
| GST_PAD_SRC, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS (RESAMPLE_CAPS) |
| ); |
| GstElement *audioresample; |
| GstCaps *caps; |
| GstStructure *structure; |
| |
| GST_DEBUG ("setup_audioresample"); |
| audioresample = gst_check_setup_element ("audioresample"); |
| |
| caps = gst_caps_from_string (RESAMPLE_CAPS); |
| structure = gst_caps_get_structure (caps, 0); |
| gst_structure_set (structure, "channels", G_TYPE_INT, channels, |
| "rate", G_TYPE_INT, inrate, "format", G_TYPE_STRING, format, |
| "channel-mask", GST_TYPE_BITMASK, mask, 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); |
| gst_pad_set_active (mysrcpad, TRUE); |
| gst_check_setup_events (mysrcpad, audioresample, caps, GST_FORMAT_TIME); |
| gst_caps_unref (caps); |
| |
| caps = gst_caps_from_string (RESAMPLE_CAPS); |
| structure = gst_caps_get_structure (caps, 0); |
| gst_structure_set (structure, "channels", G_TYPE_INT, channels, |
| "rate", G_TYPE_INT, outrate, "format", G_TYPE_STRING, format, NULL); |
| fail_unless (gst_caps_is_fixed (caps)); |
| sinktemplate = |
| gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, caps); |
| |
| mysinkpad = |
| gst_check_setup_sink_pad_from_template (audioresample, sinktemplate); |
| gst_pad_set_active (mysinkpad, TRUE); |
| /* this installs a getcaps func that will always return the caps we set |
| * later */ |
| gst_pad_use_fixed_caps (mysinkpad); |
| |
| gst_caps_unref (caps); |
| gst_object_unref (sinktemplate); |
| |
| 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; |
| GstMapInfo map; |
| gint16 *p; |
| |
| audioresample = |
| setup_audioresample (2, 0x3, inrate, outrate, GST_AUDIO_NE (S16)); |
| caps = gst_pad_get_current_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_map (inbuffer, &map, GST_MAP_WRITE); |
| p = (gint16 *) map.data; |
| |
| /* 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_buffer_unmap (inbuffer, &map); |
| |
| /* 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; |
| GstMapInfo map; |
| gint16 *p; |
| |
| GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d", |
| inrate, outrate, samples, numbuffers); |
| |
| audioresample = |
| setup_audioresample (2, 3, inrate, outrate, GST_AUDIO_NE (S16)); |
| caps = gst_pad_get_current_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_map (inbuffer, &map, GST_MAP_WRITE); |
| p = (gint16 *) map.data; |
| /* 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_buffer_unmap (inbuffer, &map); |
| |
| 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; |
| GstSegment segment; |
| |
| audioresample = setup_audioresample (1, 0, 9343, 48000, GST_AUDIO_NE (S16)); |
| caps = gst_pad_get_current_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"); |
| |
| gst_segment_init (&segment, GST_FORMAT_TIME); |
| newseg = gst_event_new_segment (&segment); |
| fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE); |
| |
| inbuffer = gst_buffer_new_and_alloc (9343 * 4); |
| gst_buffer_memset (inbuffer, 0, 0, 9343 * 4); |
| 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); |
| |
| /* 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_segment (&segment); |
| fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE); |
| |
| inbuffer = gst_buffer_new_and_alloc (9343 * 4); |
| gst_buffer_memset (inbuffer, 0, 0, 9343 * 4); |
| GST_BUFFER_DURATION (inbuffer) = GST_SECOND; |
| GST_BUFFER_TIMESTAMP (inbuffer) = 0; |
| GST_BUFFER_OFFSET (inbuffer) = 0; |
| |
| 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", "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", "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; |
| |
| #if 0 |
| 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_make_writable (gst_pad_get_current_caps (pad)); |
| |
| gst_caps_set_simple (result, |
| "rate", GST_TYPE_INT_RANGE, 48000, G_MAXINT, NULL); |
| |
| return result; |
| } |
| #endif |
| |
| 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); |
| |
| #if 0 |
| fail_unless (gst_pad_alloc_buffer_and_set_caps (mysrcpad, |
| GST_BUFFER_OFFSET_NONE, rate * 4, desired, &inbuffer) == GST_FLOW_OK); |
| #endif |
| inbuffer = gst_buffer_new_and_alloc (rate * 4); |
| gst_buffer_memset (inbuffer, 0, 0, rate * 4); |
| |
| 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; |
| GstSegment segment; |
| |
| audioresample = |
| setup_audioresample (4, 0xf, 48000, 48000, GST_AUDIO_NE (S16)); |
| |
| /* 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_current_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"); |
| |
| gst_segment_init (&segment, GST_FORMAT_TIME); |
| newseg = gst_event_new_segment (&segment); |
| 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 (const gchar * format, 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,format=%s,rate=%d,channels=2 ! audioresample quality=%d ! audio/x-raw,format=%s,rate=%d ! identity check-imperfect-timestamp=TRUE ! fakesink", |
| format, inrate, quality, format, outrate); |
| |
| 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); |
| |
| gst_bus_remove_signal_watch (bus); |
| gst_object_unref (bus); |
| |
| 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 ("S8", 44100, 48000, quality); |
| test_pipeline ("S8", 48000, 44100, quality); |
| |
| test_pipeline (GST_AUDIO_NE (S16), 44100, 48000, quality); |
| test_pipeline (GST_AUDIO_NE (S16), 48000, 44100, quality); |
| |
| test_pipeline (GST_AUDIO_NE (S24), 44100, 48000, quality); |
| test_pipeline (GST_AUDIO_NE (S24), 48000, 44100, quality); |
| |
| test_pipeline (GST_AUDIO_NE (S32), 44100, 48000, quality); |
| test_pipeline (GST_AUDIO_NE (S32), 48000, 44100, quality); |
| |
| test_pipeline (GST_AUDIO_NE (F32), 44100, 48000, quality); |
| test_pipeline (GST_AUDIO_NE (F32), 48000, 44100, quality); |
| |
| test_pipeline (GST_AUDIO_NE (F64), 44100, 48000, quality); |
| test_pipeline (GST_AUDIO_NE (F64), 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,format=" GST_AUDIO_NE (S16) ",channels=1," |
| "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_current_caps (pad); |
| GST_LOG ("current 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_get_size (buffer) / 8; |
| } |
| |
| /* Check if we have a perfectly timestamped 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_get_size (buffer) / 8, |
| "expected offset end %" G_GUINT64_FORMAT " got offset end %" |
| G_GUINT64_FORMAT, |
| GST_BUFFER_OFFSET (buffer) + gst_buffer_get_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, format=" GST_AUDIO_NE (F64) |
| ", channels=1, 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, format=" GST_AUDIO_NE (F64) |
| ", channels=1, 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_bus_remove_signal_watch (bus); |
| gst_object_unref (bus); |
| |
| gst_object_unref (pipeline); |
| |
| } GST_END_TEST; |
| |
| #define FFT_HELPERS(type,ffttag,ffttag2,scale); \ |
| static gdouble magnitude##ffttag (const GstFFT##ffttag##Complex *c) \ |
| { \ |
| gdouble mag = (gdouble) c->r * (gdouble) c->r; \ |
| mag += (gdouble) c->i * (gdouble) c->i; \ |
| mag /= scale * scale; \ |
| mag = 10.0 * log10 (mag); \ |
| return mag; \ |
| } \ |
| static gdouble find_main_frequency_spot_##ffttag (const GstFFT##ffttag##Complex *v, \ |
| int elements) \ |
| { \ |
| int i; \ |
| gdouble maxmag = -9999; \ |
| int maxidx = 0; \ |
| for (i=0; i<elements; ++i) { \ |
| gdouble mag = magnitude##ffttag (v+i); \ |
| if (mag > maxmag) { \ |
| maxmag = mag; \ |
| maxidx = i; \ |
| } \ |
| } \ |
| return maxidx / (gdouble) elements; \ |
| } \ |
| static gboolean is_zero_except_##ffttag (const GstFFT##ffttag##Complex *v, int elements, \ |
| gdouble spot) \ |
| { \ |
| int i; \ |
| for (i=0; i<elements; ++i) { \ |
| gdouble pos = i / (gdouble) elements; \ |
| gdouble mag = magnitude##ffttag (v+i); \ |
| if (fabs (pos - spot) > 0.01) { \ |
| if (mag > -55.0) { \ |
| return FALSE; \ |
| } \ |
| } \ |
| } \ |
| return TRUE; \ |
| } \ |
| static void compare_ffts_##ffttag (GstBuffer *inbuffer, GstBuffer *outbuffer) \ |
| { \ |
| GstMapInfo inmap, outmap; \ |
| int insamples, outsamples; \ |
| gdouble inspot, outspot; \ |
| GstFFT##ffttag *inctx, *outctx; \ |
| GstFFT##ffttag##Complex *in, *out; \ |
| \ |
| gst_buffer_map (inbuffer, &inmap, GST_MAP_READ); \ |
| gst_buffer_map (outbuffer, &outmap, GST_MAP_READWRITE); \ |
| \ |
| insamples = inmap.size / sizeof(type) & ~1; \ |
| outsamples = outmap.size / sizeof(type) & ~1; \ |
| inctx = gst_fft_##ffttag2##_new (insamples, FALSE); \ |
| outctx = gst_fft_##ffttag2##_new (outsamples, FALSE); \ |
| in = g_new (GstFFT##ffttag##Complex, insamples / 2 + 1); \ |
| out = g_new (GstFFT##ffttag##Complex, outsamples / 2 + 1); \ |
| \ |
| gst_fft_##ffttag2##_window (inctx, (type*)inmap.data, \ |
| GST_FFT_WINDOW_HAMMING); \ |
| gst_fft_##ffttag2##_fft (inctx, (type*)inmap.data, in); \ |
| gst_fft_##ffttag2##_window (outctx, (type*)outmap.data, \ |
| GST_FFT_WINDOW_HAMMING); \ |
| gst_fft_##ffttag2##_fft (outctx, (type*)outmap.data, out); \ |
| \ |
| inspot = find_main_frequency_spot_##ffttag (in, insamples / 2 + 1); \ |
| outspot = find_main_frequency_spot_##ffttag (out, outsamples / 2 + 1); \ |
| GST_LOG ("Spots are %.3f and %.3f", inspot, outspot); \ |
| fail_unless (fabs (outspot - inspot) < 0.05); \ |
| fail_unless (is_zero_except_##ffttag (in, insamples / 2 + 1, inspot)); \ |
| fail_unless (is_zero_except_##ffttag (out, outsamples / 2 + 1, outspot)); \ |
| \ |
| gst_buffer_unmap (inbuffer, &inmap); \ |
| gst_buffer_unmap (outbuffer, &outmap); \ |
| \ |
| gst_fft_##ffttag2##_free (inctx); \ |
| gst_fft_##ffttag2##_free (outctx); \ |
| g_free (in); \ |
| g_free (out); \ |
| } |
| FFT_HELPERS (float, F32, f32, 2048.0f); |
| FFT_HELPERS (double, F64, f64, 2048.0); |
| FFT_HELPERS (gint16, S16, s16, 32767.0); |
| FFT_HELPERS (gint32, S32, s32, 2147483647.0); |
| |
| #define FILL_BUFFER(type, desc, value); \ |
| static void init_##type##_##desc (GstBuffer *buffer) \ |
| { \ |
| GstMapInfo map; \ |
| type *ptr; \ |
| int i, nsamples; \ |
| gst_buffer_map (buffer, &map, GST_MAP_WRITE); \ |
| ptr = (type *)map.data; \ |
| nsamples = map.size / sizeof (type); \ |
| for (i = 0; i < nsamples; ++i) { \ |
| *ptr++ = value; \ |
| } \ |
| gst_buffer_unmap (buffer, &map); \ |
| } |
| |
| FILL_BUFFER (float, silence, 0.0f); |
| FILL_BUFFER (double, silence, 0.0); |
| FILL_BUFFER (gint16, silence, 0); |
| FILL_BUFFER (gint32, silence, 0); |
| FILL_BUFFER (float, sine, sinf (i * 0.01f)); |
| FILL_BUFFER (float, sine2, sinf (i * 1.8f)); |
| FILL_BUFFER (double, sine, sin (i * 0.01)); |
| FILL_BUFFER (double, sine2, sin (i * 1.8)); |
| FILL_BUFFER (gint16, sine, (gint16) (32767 * sinf (i * 0.01f))); |
| FILL_BUFFER (gint16, sine2, (gint16) (32767 * sinf (i * 1.8f))); |
| FILL_BUFFER (gint32, sine, (gint32) (2147483647 * sinf (i * 0.01f))); |
| FILL_BUFFER (gint32, sine2, (gint32) (2147483647 * sinf (i * 1.8f))); |
| |
| static void |
| run_fft_pipeline (int inrate, int outrate, int quality, int width, |
| const gchar * format, void (*init) (GstBuffer *), |
| void (*compare_ffts) (GstBuffer *, GstBuffer *)) |
| { |
| GstElement *audioresample; |
| GstBuffer *inbuffer, *outbuffer; |
| const int nsamples = 2048; |
| |
| audioresample = setup_audioresample (1, 0, inrate, outrate, format); |
| fail_unless (audioresample != NULL); |
| g_object_set (audioresample, "quality", quality, NULL); |
| |
| fail_unless (gst_element_set_state (audioresample, |
| GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, |
| "could not set to playing"); |
| |
| inbuffer = gst_buffer_new_and_alloc (nsamples * width / 8); |
| GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (nsamples, inrate); |
| GST_BUFFER_TIMESTAMP (inbuffer) = 0; |
| |
| (*init) (inbuffer); |
| |
| gst_buffer_ref (inbuffer); |
| /* 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); |
| /* retrieve out buffer */ |
| fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); |
| |
| fail_unless (gst_element_set_state (audioresample, |
| GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to null"); |
| |
| if (inbuffer == outbuffer) |
| gst_buffer_unref (inbuffer); |
| |
| (*compare_ffts) (inbuffer, outbuffer); |
| |
| /* cleanup */ |
| cleanup_audioresample (audioresample); |
| } |
| |
| GST_START_TEST (test_fft) |
| { |
| int quality; |
| size_t f0, f1; |
| static const int frequencies[] = |
| { 8000, 16000, 44100, 48000, 128000, 12345, 54321 }; |
| |
| /* audioresample uses a mixed float/double code path for floats with quality>8, make sure we test it */ |
| for (quality = 0; quality <= 10; quality += 5) { |
| for (f0 = 0; f0 < G_N_ELEMENTS (frequencies); ++f0) { |
| for (f1 = 0; f1 < G_N_ELEMENTS (frequencies); ++f1) { |
| run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, |
| GST_AUDIO_NE (F32), &init_float_silence, &compare_ffts_F32); |
| run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, |
| GST_AUDIO_NE (F32), &init_float_sine, &compare_ffts_F32); |
| run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, |
| GST_AUDIO_NE (F32), &init_float_sine2, &compare_ffts_F32); |
| run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64, |
| GST_AUDIO_NE (F64), &init_double_silence, &compare_ffts_F64); |
| run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64, |
| GST_AUDIO_NE (F64), &init_double_sine, &compare_ffts_F64); |
| run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64, |
| GST_AUDIO_NE (F64), &init_double_sine2, &compare_ffts_F64); |
| run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16, |
| GST_AUDIO_NE (S16), &init_gint16_silence, &compare_ffts_S16); |
| run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16, |
| GST_AUDIO_NE (S16), &init_gint16_sine, &compare_ffts_S16); |
| run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16, |
| GST_AUDIO_NE (S16), &init_gint16_sine2, &compare_ffts_S16); |
| run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, |
| GST_AUDIO_NE (S32), &init_gint32_silence, &compare_ffts_S32); |
| run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, |
| GST_AUDIO_NE (S32), &init_gint32_sine, &compare_ffts_S32); |
| run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, |
| GST_AUDIO_NE (S32), &init_gint32_sine2, &compare_ffts_S32); |
| } |
| } |
| } |
| } |
| |
| 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); |
| tcase_add_test (tc_chain, test_fft); |
| |
| #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); |
