| /* 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. | |
| */ | |
| #ifdef HAVE_CONFIG_H | |
| #include "config.h" | |
| #endif | |
| #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; | |
| static void | |
| live_switch_push (gint pts, gint rate, GstCaps * caps) | |
| { | |
| GstBuffer *inbuffer; | |
| GstCaps *desired; | |
| desired = gst_caps_copy (caps); | |
| gst_caps_set_simple (desired, "rate", G_TYPE_INT, rate, NULL); | |
| gst_pad_set_caps (mysrcpad, desired); | |
| inbuffer = gst_buffer_new_and_alloc (rate * 4 * 2); | |
| gst_buffer_memset (inbuffer, 0, 0, rate * 4 * 2); | |
| GST_BUFFER_DURATION (inbuffer) = GST_SECOND; | |
| GST_BUFFER_TIMESTAMP (inbuffer) = pts * GST_SECOND; | |
| GST_BUFFER_OFFSET (inbuffer) = 0; | |
| GST_BUFFER_OFFSET_END (inbuffer) = rate - 1; | |
| /* pushing gives away my reference ... */ | |
| fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK); | |
| /* ... but it ends up being collected on the global buffer list */ | |
| gst_caps_unref (desired); | |
| } | |
| GST_START_TEST (test_live_switch) | |
| { | |
| GstElement *audioresample; | |
| GstEvent *newseg; | |
| GstCaps *caps; | |
| GstSegment segment; | |
| GList *l; | |
| guint i; | |
| audioresample = | |
| setup_audioresample (4, 0xf, 48000, 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); | |
| /* downstream can accept the requested rate */ | |
| live_switch_push (0, 48000, caps); | |
| /* buffer is directly passed through */ | |
| fail_unless_equals_int (g_list_length (buffers), 1); | |
| /* Downstream can never accept this rate */ | |
| live_switch_push (1, 40000, caps); | |
| /* one additional buffer is provided with the new sample rate */ | |
| fail_unless_equals_int (g_list_length (buffers), 2); | |
| /* Downstream can never accept this rate */ | |
| live_switch_push (2, 50000, caps); | |
| /* two additional buffers are provided. One is the drained remainder of | |
| * the previous sample rate, the second is the buffer with the new sample | |
| * rate */ | |
| fail_unless_equals_int (g_list_length (buffers), 4); | |
| /* Send EOS to drain the remaining samples */ | |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); | |
| fail_unless_equals_int (g_list_length (buffers), 5); | |
| /* Now test that each buffer has the expected samples. We simply check this | |
| * by checking whether the timestamps, durations and sizes are matching */ | |
| for (l = buffers, i = 0; l; l = l->next, i++) { | |
| GstBuffer *buffer = GST_BUFFER (l->data); | |
| switch (i) { | |
| case 0: | |
| fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 0 * GST_SECOND); | |
| fail_unless_equals_uint64 (GST_BUFFER_DURATION (buffer), | |
| 1 * GST_SECOND); | |
| fail_unless_equals_int (gst_buffer_get_size (buffer), 48000 * 4 * 2); | |
| break; | |
| case 1: | |
| fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 1 * GST_SECOND); | |
| fail_unless_equals_int (gst_buffer_get_size (buffer), 47961 * 4 * 2); | |
| break; | |
| case 2: | |
| fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) + | |
| GST_BUFFER_DURATION (buffer), 2 * GST_SECOND, | |
| GST_SECOND / 48000 + 1)); | |
| fail_unless_equals_int (gst_buffer_get_size (buffer), 38 * 4 * 2); | |
| break; | |
| case 3: | |
| fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 2 * GST_SECOND); | |
| fail_unless_equals_int (gst_buffer_get_size (buffer), 47969 * 4 * 2); | |
| break; | |
| case 4: | |
| fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) + | |
| GST_BUFFER_DURATION (buffer), 3 * GST_SECOND, | |
| GST_SECOND / 48000 + 1)); | |
| fail_unless_equals_int (gst_buffer_get_size (buffer), 30 * 4 * 2); | |
| break; | |
| default: | |
| g_assert_not_reached (); | |
| break; | |
| } | |
| gst_buffer_unref (buffer); | |
| } | |
| g_list_free (buffers); | |
| buffers = NULL; | |
| cleanup_audioresample (audioresample); | |
| gst_caps_unref (caps); | |
| } | |
| GST_END_TEST; | |
| static gint current_rate = 0; | |
| static gboolean | |
| live_switch_sink_query (GstPad * pad, GstObject * parent, GstQuery * query) | |
| { | |
| switch (GST_QUERY_TYPE (query)) { | |
| case GST_QUERY_ACCEPT_CAPS:{ | |
| GstCaps *acceptable_caps; | |
| GstCaps *caps; | |
| acceptable_caps = gst_pad_get_current_caps (mysrcpad); | |
| acceptable_caps = gst_caps_make_writable (acceptable_caps); | |
| gst_caps_set_simple (acceptable_caps, "rate", G_TYPE_INT, current_rate, | |
| NULL); | |
| gst_query_parse_accept_caps (query, &caps); | |
| gst_query_set_accept_caps_result (query, gst_caps_can_intersect (caps, | |
| acceptable_caps)); | |
| gst_caps_unref (acceptable_caps); | |
| return TRUE; | |
| } | |
| case GST_QUERY_CAPS:{ | |
| GstCaps *acceptable_caps; | |
| GstCaps *filter; | |
| GstCaps *caps; | |
| acceptable_caps = gst_pad_get_current_caps (mysrcpad); | |
| acceptable_caps = gst_caps_make_writable (acceptable_caps); | |
| gst_caps_set_simple (acceptable_caps, "rate", G_TYPE_INT, current_rate, | |
| NULL); | |
| gst_query_parse_caps (query, &filter); | |
| if (filter) | |
| caps = | |
| gst_caps_intersect_full (filter, acceptable_caps, | |
| GST_CAPS_INTERSECT_FIRST); | |
| else | |
| caps = gst_caps_ref (acceptable_caps); | |
| gst_query_set_caps_result (query, caps); | |
| gst_caps_unref (caps); | |
| gst_caps_unref (acceptable_caps); | |
| return TRUE; | |
| } | |
| default: | |
| return gst_pad_query_default (pad, parent, query); | |
| } | |
| } | |
| static void | |
| live_switch_push_downstream (gint pts, gint rate) | |
| { | |
| GstBuffer *inbuffer; | |
| current_rate = rate; | |
| gst_pad_push_event (mysinkpad, gst_event_new_reconfigure ()); | |
| inbuffer = gst_buffer_new_and_alloc (48000 * 4 * 2); | |
| gst_buffer_memset (inbuffer, 0, 0, 48000 * 4 * 2); | |
| GST_BUFFER_DURATION (inbuffer) = GST_SECOND; | |
| GST_BUFFER_TIMESTAMP (inbuffer) = pts * GST_SECOND; | |
| GST_BUFFER_OFFSET (inbuffer) = 0; | |
| GST_BUFFER_OFFSET_END (inbuffer) = 47999; | |
| /* pushing gives away my reference ... */ | |
| fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK); | |
| /* ... but it ends up being collected on the global buffer list */ | |
| } | |
| GST_START_TEST (test_live_switch_downstream) | |
| { | |
| GstElement *audioresample; | |
| GstEvent *newseg; | |
| GstCaps *caps; | |
| GstSegment segment; | |
| GList *l; | |
| guint i; | |
| audioresample = | |
| setup_audioresample (4, 0xf, 48000, 48000, GST_AUDIO_NE (S16)); | |
| gst_pad_set_query_function (mysinkpad, live_switch_sink_query); | |
| 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); | |
| /* buffer is directly passed through */ | |
| live_switch_push_downstream (0, 48000); | |
| fail_unless_equals_int (g_list_length (buffers), 1); | |
| /* Reconfigure downstream to 40000 Hz */ | |
| live_switch_push_downstream (1, 40000); | |
| /* one additional buffer is provided with the new sample rate */ | |
| fail_unless_equals_int (g_list_length (buffers), 2); | |
| /* Reconfigure downstream to 50000 Hz */ | |
| live_switch_push_downstream (2, 50000); | |
| /* two additional buffers are provided. One is the drained remainder of | |
| * the previous sample rate, the second is the buffer with the new sample | |
| * rate */ | |
| fail_unless_equals_int (g_list_length (buffers), 4); | |
| /* Send EOS to drain the remaining samples */ | |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); | |
| fail_unless_equals_int (g_list_length (buffers), 5); | |
| /* Now test that each buffer has the expected samples. We simply check this | |
| * by checking whether the timestamps, durations and sizes are matching */ | |
| for (l = buffers, i = 0; l; l = l->next, i++) { | |
| GstBuffer *buffer = GST_BUFFER (l->data); | |
| switch (i) { | |
| case 0: | |
| fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 0 * GST_SECOND); | |
| fail_unless_equals_uint64 (GST_BUFFER_DURATION (buffer), | |
| 1 * GST_SECOND); | |
| fail_unless_equals_int (gst_buffer_get_size (buffer), 48000 * 4 * 2); | |
| break; | |
| case 1: | |
| fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 1 * GST_SECOND); | |
| fail_unless_equals_int (gst_buffer_get_size (buffer), 39966 * 4 * 2); | |
| break; | |
| case 2: | |
| fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) + | |
| GST_BUFFER_DURATION (buffer), 2 * GST_SECOND, | |
| GST_SECOND / 40000 + 1)); | |
| fail_unless_equals_int (gst_buffer_get_size (buffer), 34 * 4 * 2); | |
| break; | |
| case 3: | |
| fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 2 * GST_SECOND); | |
| fail_unless_equals_int (gst_buffer_get_size (buffer), 49966 * 4 * 2); | |
| break; | |
| case 4: | |
| fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) + | |
| GST_BUFFER_DURATION (buffer), 3 * GST_SECOND, | |
| GST_SECOND / 50000 + 1)); | |
| fail_unless_equals_int (gst_buffer_get_size (buffer), 33 * 4 * 2); | |
| break; | |
| default: | |
| g_assert_not_reached (); | |
| break; | |
| } | |
| gst_buffer_unref (buffer); | |
| } | |
| g_list_free (buffers); | |
| buffers = NULL; | |
| 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) (2147483647.0 * sin (i * 0.01))); | |
| FILL_BUFFER (gint32, sine2, (gint32) (2147483647.0 * sin (i * 1.8))); | |
| 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_live_switch_downstream); | |
| 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); |