tests: add interactive test for accurate seeking

For some audio formats.

https://bugzilla.gnome.org/show_bug.cgi?id=655276

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;
+}