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-rw-r--r--tests/icles/.gitignore1
-rw-r--r--tests/icles/Makefile.am14
-rw-r--r--tests/icles/test-accurate-seek.c275
3 files changed, 288 insertions, 2 deletions
diff --git a/tests/icles/.gitignore b/tests/icles/.gitignore
index 0b5704b32..2810ce2f0 100644
--- a/tests/icles/.gitignore
+++ b/tests/icles/.gitignore
@@ -1,5 +1,6 @@
equalizer-test
gdkpixbufsink-test
+test-accurate-seek
test-oss4
ximagesrc-test
v4l2src-test
diff --git a/tests/icles/Makefile.am b/tests/icles/Makefile.am
index 9ff34a9c5..a4c9f3591 100644
--- a/tests/icles/Makefile.am
+++ b/tests/icles/Makefile.am
@@ -39,6 +39,12 @@ equalizer_test_SOURCES = equalizer-test.c
equalizer_test_CFLAGS = $(GST_CFLAGS)
equalizer_test_LDADD = $(GST_LIBS)
+test_accurate_seek_SOURCES = test-accurate-seek.c
+test_accurate_seek_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) \
+ $(GST_BASE_CFLAGS) $(GST_CFLAGS)
+test_accurate_seek_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstapp-$(GST_API_VERSION) \
+ $(GST_BASE_LIBS) $(GST_LIBS)
+
videocrop_test_SOURCES = videocrop-test.c
videocrop_test_CFLAGS = $(GST_CFLAGS)
videocrop_test_LDADD = $(GST_LIBS)
@@ -51,5 +57,9 @@ videocrop2_test_SOURCES = videocrop2-test.c
videocrop2_test_CFLAGS = $(GST_CFLAGS)
videocrop2_test_LDADD = $(GST_LIBS)
-noinst_PROGRAMS = $(GTK_TESTS) $(OSS4_TESTS) $(V4L2_TESTS) $(X_TESTS) equalizer-test videocrop-test videobox-test videocrop2-test
-
+noinst_PROGRAMS = $(GTK_TESTS) $(OSS4_TESTS) $(V4L2_TESTS) $(X_TESTS) \
+ equalizer-test \
+ test-accurate-seek \
+ videocrop-test \
+ videobox-test \
+ videocrop2-test
diff --git a/tests/icles/test-accurate-seek.c b/tests/icles/test-accurate-seek.c
new file mode 100644
index 000000000..9e9da104e
--- /dev/null
+++ b/tests/icles/test-accurate-seek.c
@@ -0,0 +1,275 @@
+/* GStreamer interactive test for accurate seeking
+ * Copyright (C) 2014 Tim-Philipp Müller <tim centricular com>
+ *
+ * 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.
+ *
+ * Based on python script by Kibeom Kim <kkb110@gmail.com>
+ */
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#define _GNU_SOURCE /* for memmem */
+#include <string.h>
+
+#include <gst/gst.h>
+#include <gst/base/base.h>
+#include <gst/audio/audio.h>
+#include <gst/app/app.h>
+
+#define SAMPLE_FREQ 44100
+
+static GstClockTime
+sample_to_nanotime (guint sample)
+{
+ return (guint64) ((1.0 * sample * GST_SECOND / SAMPLE_FREQ) + 0.5);
+}
+
+static guint
+nanotime_to_sample (GstClockTime nanotime)
+{
+ return nanotime * SAMPLE_FREQ / GST_SECOND;
+}
+
+static GstBuffer *
+generate_test_data (guint N)
+{
+ gint16 *left, *right, *stereo;
+ guint largeN, i, j;
+
+ /* 32767 = (2 ** 15) - 1 */
+ /* 32768 = (2 ** 15) */
+ largeN = ((N + 32767) / 32768) * 32768;
+ left = g_new0 (gint16, largeN);
+ right = g_new0 (gint16, largeN);
+ stereo = g_new0 (gint16, 2 * largeN);
+
+ for (i = 0; i < (largeN / 32768); ++i) {
+ gint c = 0;
+
+ for (j = i * 32768; j < ((i + 1) * 32768); ++j) {
+ left[j] = i;
+
+ if (i % 2 == 0) {
+ right[j] = c;
+ } else {
+ right[j] = 32767 - c;
+ }
+ ++c;
+ }
+ }
+
+ /* could just fill stereo directly from the start, but keeping original code for now */
+ for (i = 0; i < largeN; ++i) {
+ stereo[(2 * i) + 0] = left[i];
+ stereo[(2 * i) + 1] = right[i];
+ }
+ g_free (left);
+ g_free (right);
+
+ return gst_buffer_new_wrapped (stereo, 2 * largeN * sizeof (gint16));
+}
+
+static void
+generate_test_sound (const gchar * fn, const gchar * launch_string,
+ guint num_samples)
+{
+ GstElement *pipeline, *src, *parse, *enc_bin, *sink;
+ GstFlowReturn flow;
+ GstMessage *msg;
+ GstBuffer *buf;
+ GstCaps *caps;
+
+ pipeline = gst_pipeline_new (NULL);
+
+ src = gst_element_factory_make ("appsrc", NULL);
+
+ caps = gst_caps_new_simple ("audio/x-raw",
+ "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
+ "rate", G_TYPE_INT, SAMPLE_FREQ, "channels", G_TYPE_INT, 2,
+ "layout", G_TYPE_STRING, "interleaved",
+ "channel-mask", GST_TYPE_BITMASK, (guint64) 3, NULL);
+ g_object_set (src, "caps", caps, "format", GST_FORMAT_TIME, NULL);
+ gst_base_src_set_format (GST_BASE_SRC (src), GST_FORMAT_TIME);
+ gst_caps_unref (caps);
+
+ /* audioparse to put proper timestamps on buffers for us, without which
+ * vorbisenc in particular is unhappy (or oggmux, rather) */
+ parse = gst_element_factory_make ("audioparse", NULL);
+ if (parse != NULL) {
+ g_object_set (parse, "use-sink-caps", TRUE, NULL);
+ } else {
+ parse = gst_element_factory_make ("identity", NULL);
+ g_warning ("audioparse element not available, vorbis/ogg might not work\n");
+ }
+
+ enc_bin = gst_parse_bin_from_description (launch_string, TRUE, NULL);
+
+ sink = gst_element_factory_make ("filesink", NULL);
+ g_object_set (sink, "location", fn, NULL);
+
+ gst_bin_add_many (GST_BIN (pipeline), src, parse, enc_bin, sink, NULL);
+
+ gst_element_link_many (src, parse, enc_bin, sink, NULL);
+
+ gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+ buf = generate_test_data (num_samples);
+ flow = gst_app_src_push_buffer (GST_APP_SRC (src), buf);
+ g_assert (flow == GST_FLOW_OK);
+
+ gst_app_src_end_of_stream (GST_APP_SRC (src));
+
+ /*g_print ("generating test sound %s, waiting for EOS..\n", fn); */
+
+ msg = gst_bus_timed_pop_filtered (GST_ELEMENT_BUS (pipeline),
+ GST_CLOCK_TIME_NONE, GST_MESSAGE_EOS | GST_MESSAGE_ERROR);
+
+ g_assert (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_EOS);
+ gst_message_unref (msg);
+
+ gst_element_set_state (pipeline, GST_STATE_NULL);
+ gst_object_unref (pipeline);
+
+ /* g_print ("Done %s\n", fn); */
+}
+
+static void
+test_seek_FORMAT_TIME_by_sample (const gchar * fn, GList * seek_positions)
+{
+ GstElement *pipeline, *src, *sink;
+ GstAdapter *adapter;
+ GstSample *sample;
+ GstCaps *caps;
+ gconstpointer answer;
+ guint answer_size;
+
+ pipeline = gst_parse_launch ("filesrc name=src ! decodebin ! "
+ "audioconvert dithering=0 ! appsink name=sink", NULL);
+
+ src = gst_bin_get_by_name (GST_BIN (pipeline), "src");
+ g_object_set (src, "location", fn, NULL);
+ gst_object_unref (src);
+
+ sink = gst_bin_get_by_name (GST_BIN (pipeline), "sink");
+ caps = gst_caps_new_simple ("audio/x-raw",
+ "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
+ "rate", G_TYPE_INT, SAMPLE_FREQ, "channels", G_TYPE_INT, 2, NULL);
+ g_object_set (sink, "caps", caps, "sync", FALSE, NULL);
+ gst_caps_unref (caps);
+
+ gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+ /* wait for preroll, so we can seek */
+ gst_bus_timed_pop_filtered (GST_ELEMENT_BUS (pipeline), GST_CLOCK_TIME_NONE,
+ GST_MESSAGE_ASYNC_DONE);
+
+ /* first, read entire file to end */
+ adapter = gst_adapter_new ();
+ while ((sample = gst_app_sink_pull_sample (GST_APP_SINK (sink)))) {
+ gst_adapter_push (adapter, gst_buffer_ref (gst_sample_get_buffer (sample)));
+ gst_sample_unref (sample);
+ }
+ answer_size = gst_adapter_available (adapter);
+ answer = gst_adapter_map (adapter, answer_size);
+ /* g_print ("%s: read %u bytes\n", fn, answer_size); */
+
+ g_print ("%10s\t%10s\t%10s\n", "requested", "sample per ts", "actual(data)");
+
+ while (seek_positions != NULL) {
+ gconstpointer found;
+ GstMapInfo map;
+ GstBuffer *buf;
+ gboolean ret;
+ guint actual_position, buffer_timestamp_position;
+ guint seek_sample;
+
+ seek_sample = GPOINTER_TO_UINT (seek_positions->data);
+
+ ret = gst_element_seek_simple (pipeline, GST_FORMAT_TIME,
+ GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE,
+ sample_to_nanotime (seek_sample));
+
+ g_assert (ret);
+
+ sample = gst_app_sink_pull_sample (GST_APP_SINK (sink));
+
+ buf = gst_sample_get_buffer (sample);
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ found = memmem (answer, answer_size, map.data, map.size);
+ gst_buffer_unmap (buf, &map);
+
+ g_assert (found != NULL);
+ actual_position = ((goffset) ((guint8 *) found - (guint8 *) answer)) / 4;
+ buffer_timestamp_position = nanotime_to_sample (GST_BUFFER_PTS (buf));
+ g_print ("%10u\t%10u\t%10u\n", seek_sample, buffer_timestamp_position,
+ actual_position);
+ gst_sample_unref (sample);
+
+ seek_positions = seek_positions->next;
+ }
+
+ gst_element_set_state (pipeline, GST_STATE_NULL);
+ gst_object_unref (sink);
+ gst_object_unref (pipeline);
+ g_object_unref (adapter);
+}
+
+static GList *
+create_test_samples (guint from, guint to, guint step)
+{
+ GQueue q = G_QUEUE_INIT;
+ guint i;
+
+ for (i = from; i < to; i += step)
+ g_queue_push_tail (&q, GUINT_TO_POINTER (i));
+
+ return q.head;
+}
+
+#define SECS 10
+
+int
+main (int argc, char **argv)
+{
+ GList *test_samples;
+
+ gst_init (&argc, &argv);
+
+ test_samples = create_test_samples (SAMPLE_FREQ, SAMPLE_FREQ * 2, 5000);
+
+ g_print ("\nwav:\n");
+ generate_test_sound ("test.wav", "wavenc", SAMPLE_FREQ * SECS);
+ test_seek_FORMAT_TIME_by_sample ("test.wav", test_samples);
+
+ g_print ("\nflac:\n");
+ generate_test_sound ("test.flac", "flacenc", SAMPLE_FREQ * SECS);
+ test_seek_FORMAT_TIME_by_sample ("test.flac", test_samples);
+
+ g_print ("\nogg:\n");
+ generate_test_sound ("test.ogg",
+ "audioconvert dithering=0 ! vorbisenc quality=1 ! oggmux",
+ SAMPLE_FREQ * SECS);
+ test_seek_FORMAT_TIME_by_sample ("test.ogg", test_samples);
+
+ g_print ("\nmp3:\n");
+ generate_test_sound ("test.mp3", "lamemp3enc bitrate=320",
+ SAMPLE_FREQ * SECS);
+ test_seek_FORMAT_TIME_by_sample ("test.mp3", test_samples);
+
+ g_list_free (test_samples);
+ return 0;
+}
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-24 07:06:05 +0000