tests/icles/test-accurate-seek.c - mtk-gst-plugins-good-debian - Git at Google

 /* 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 gst_util_uint64_scale_round (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;
 }
	/* 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 gst_util_uint64_scale_round (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;
	}