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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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299 lines
8.6 KiB
C
299 lines
8.6 KiB
C
/* GStreamer interactive test for accurate seeking
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* Copyright (C) 2014 Tim-Philipp Müller <tim centricular com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*
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* Based on python script by Kibeom Kim <kkb110@gmail.com>
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <string.h>
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#include <gst/gst.h>
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#include <gst/base/base.h>
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#include <gst/audio/audio.h>
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#include <gst/app/app.h>
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#define SAMPLE_FREQ 44100
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static const guint8 *
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_memmem (const guint8 * haystack, gsize hlen, const guint8 * needle, gsize nlen)
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{
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const guint8 *p = haystack;
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int needle_first;
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gsize plen = hlen;
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if (!nlen)
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return NULL;
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needle_first = *(unsigned char *) needle;
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while (plen >= nlen && (p = memchr (p, needle_first, plen - nlen + 1))) {
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if (!memcmp (p, needle, nlen))
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return (guint8 *) p;
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p++;
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plen = hlen - (p - haystack);
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}
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return NULL;
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}
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static GstClockTime
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sample_to_nanotime (guint sample)
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{
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return (guint64) ((1.0 * sample * GST_SECOND / SAMPLE_FREQ) + 0.5);
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}
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static guint
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nanotime_to_sample (GstClockTime nanotime)
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{
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return gst_util_uint64_scale_round (nanotime, SAMPLE_FREQ, GST_SECOND);
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}
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static GstBuffer *
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generate_test_data (guint N)
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{
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gint16 *left, *right, *stereo;
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guint largeN, i, j;
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/* 32767 = (2 ** 15) - 1 */
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/* 32768 = (2 ** 15) */
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largeN = ((N + 32767) / 32768) * 32768;
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left = g_new0 (gint16, largeN);
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right = g_new0 (gint16, largeN);
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stereo = g_new0 (gint16, 2 * largeN);
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for (i = 0; i < (largeN / 32768); ++i) {
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gint c = 0;
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for (j = i * 32768; j < ((i + 1) * 32768); ++j) {
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left[j] = i;
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if (i % 2 == 0) {
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right[j] = c;
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} else {
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right[j] = 32767 - c;
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}
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++c;
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}
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}
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/* could just fill stereo directly from the start, but keeping original code for now */
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for (i = 0; i < largeN; ++i) {
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stereo[(2 * i) + 0] = left[i];
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stereo[(2 * i) + 1] = right[i];
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}
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g_free (left);
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g_free (right);
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return gst_buffer_new_wrapped (stereo, 2 * largeN * sizeof (gint16));
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}
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static void
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generate_test_sound (const gchar * fn, const gchar * launch_string,
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guint num_samples)
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{
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GstElement *pipeline, *src, *parse, *enc_bin, *sink;
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GstFlowReturn flow;
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GstMessage *msg;
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GstBuffer *buf;
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GstCaps *caps;
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pipeline = gst_pipeline_new (NULL);
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src = gst_element_factory_make ("appsrc", NULL);
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caps = gst_caps_new_simple ("audio/x-raw",
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"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
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"rate", G_TYPE_INT, SAMPLE_FREQ, "channels", G_TYPE_INT, 2,
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"layout", G_TYPE_STRING, "interleaved",
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"channel-mask", GST_TYPE_BITMASK, (guint64) 3, NULL);
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g_object_set (src, "caps", caps, "format", GST_FORMAT_TIME, NULL);
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gst_base_src_set_format (GST_BASE_SRC (src), GST_FORMAT_TIME);
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gst_caps_unref (caps);
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/* audioparse to put proper timestamps on buffers for us, without which
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* vorbisenc in particular is unhappy (or oggmux, rather) */
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parse = gst_element_factory_make ("audioparse", NULL);
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if (parse != NULL) {
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g_object_set (parse, "use-sink-caps", TRUE, NULL);
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} else {
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parse = gst_element_factory_make ("identity", NULL);
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g_warning ("audioparse element not available, vorbis/ogg might not work\n");
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}
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enc_bin = gst_parse_bin_from_description (launch_string, TRUE, NULL);
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sink = gst_element_factory_make ("filesink", NULL);
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g_object_set (sink, "location", fn, NULL);
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gst_bin_add_many (GST_BIN (pipeline), src, parse, enc_bin, sink, NULL);
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gst_element_link_many (src, parse, enc_bin, sink, NULL);
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gst_element_set_state (pipeline, GST_STATE_PLAYING);
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buf = generate_test_data (num_samples);
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flow = gst_app_src_push_buffer (GST_APP_SRC (src), buf);
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g_assert (flow == GST_FLOW_OK);
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gst_app_src_end_of_stream (GST_APP_SRC (src));
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/*g_print ("generating test sound %s, waiting for EOS..\n", fn); */
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msg = gst_bus_timed_pop_filtered (GST_ELEMENT_BUS (pipeline),
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GST_CLOCK_TIME_NONE, GST_MESSAGE_EOS | GST_MESSAGE_ERROR);
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g_assert (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_EOS);
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gst_message_unref (msg);
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gst_element_set_state (pipeline, GST_STATE_NULL);
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gst_object_unref (pipeline);
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/* g_print ("Done %s\n", fn); */
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}
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static void
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test_seek_FORMAT_TIME_by_sample (const gchar * fn, GList * seek_positions)
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{
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GstElement *pipeline, *src, *sink;
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GstAdapter *adapter;
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GstSample *sample;
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GstCaps *caps;
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gconstpointer answer;
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guint answer_size;
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pipeline = gst_parse_launch ("filesrc name=src ! decodebin ! "
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"audioconvert dithering=0 ! appsink name=sink", NULL);
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src = gst_bin_get_by_name (GST_BIN (pipeline), "src");
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g_object_set (src, "location", fn, NULL);
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gst_object_unref (src);
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sink = gst_bin_get_by_name (GST_BIN (pipeline), "sink");
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caps = gst_caps_new_simple ("audio/x-raw",
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"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
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"rate", G_TYPE_INT, SAMPLE_FREQ, "channels", G_TYPE_INT, 2, NULL);
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g_object_set (sink, "caps", caps, "sync", FALSE, NULL);
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gst_caps_unref (caps);
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gst_element_set_state (pipeline, GST_STATE_PLAYING);
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/* wait for preroll, so we can seek */
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gst_bus_timed_pop_filtered (GST_ELEMENT_BUS (pipeline), GST_CLOCK_TIME_NONE,
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GST_MESSAGE_ASYNC_DONE);
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/* first, read entire file to end */
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adapter = gst_adapter_new ();
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while ((sample = gst_app_sink_pull_sample (GST_APP_SINK (sink)))) {
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gst_adapter_push (adapter, gst_buffer_ref (gst_sample_get_buffer (sample)));
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gst_sample_unref (sample);
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}
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answer_size = gst_adapter_available (adapter);
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answer = gst_adapter_map (adapter, answer_size);
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/* g_print ("%s: read %u bytes\n", fn, answer_size); */
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g_print ("%10s\t%10s\t%10s\n", "requested", "sample per ts", "actual(data)");
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while (seek_positions != NULL) {
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gconstpointer found;
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GstMapInfo map;
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GstBuffer *buf;
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gboolean ret;
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guint actual_position, buffer_timestamp_position;
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guint seek_sample;
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seek_sample = GPOINTER_TO_UINT (seek_positions->data);
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ret = gst_element_seek_simple (pipeline, GST_FORMAT_TIME,
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GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE,
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sample_to_nanotime (seek_sample));
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g_assert (ret);
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sample = gst_app_sink_pull_sample (GST_APP_SINK (sink));
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buf = gst_sample_get_buffer (sample);
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gst_buffer_map (buf, &map, GST_MAP_READ);
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GST_MEMDUMP ("answer", answer, answer_size);
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GST_MEMDUMP ("buffer", map.data, map.size);
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found = _memmem (answer, answer_size, map.data, map.size);
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gst_buffer_unmap (buf, &map);
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g_assert (found != NULL);
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actual_position = ((goffset) ((guint8 *) found - (guint8 *) answer)) / 4;
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buffer_timestamp_position = nanotime_to_sample (GST_BUFFER_PTS (buf));
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g_print ("%10u\t%10u\t%10u\n", seek_sample, buffer_timestamp_position,
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actual_position);
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gst_sample_unref (sample);
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seek_positions = seek_positions->next;
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}
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gst_element_set_state (pipeline, GST_STATE_NULL);
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gst_object_unref (sink);
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gst_object_unref (pipeline);
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g_object_unref (adapter);
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}
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static GList *
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create_test_samples (guint from, guint to, guint step)
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{
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GQueue q = G_QUEUE_INIT;
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guint i;
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for (i = from; i < to; i += step)
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g_queue_push_tail (&q, GUINT_TO_POINTER (i));
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return q.head;
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}
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#define SECS 10
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int
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main (int argc, char **argv)
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{
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GList *test_samples;
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gst_init (&argc, &argv);
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test_samples = create_test_samples (SAMPLE_FREQ, SAMPLE_FREQ * 2, 5000);
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g_print ("\nwav:\n");
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generate_test_sound ("test.wav", "wavenc", SAMPLE_FREQ * SECS);
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test_seek_FORMAT_TIME_by_sample ("test.wav", test_samples);
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g_print ("\nflac:\n");
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generate_test_sound ("test.flac", "flacenc", SAMPLE_FREQ * SECS);
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test_seek_FORMAT_TIME_by_sample ("test.flac", test_samples);
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g_print ("\nogg:\n");
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generate_test_sound ("test.ogg",
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"audioconvert dithering=0 ! vorbisenc quality=1 ! oggmux",
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SAMPLE_FREQ * SECS);
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test_seek_FORMAT_TIME_by_sample ("test.ogg", test_samples);
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g_print ("\nmp3:\n");
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generate_test_sound ("test.mp3", "lamemp3enc bitrate=320",
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SAMPLE_FREQ * SECS);
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test_seek_FORMAT_TIME_by_sample ("test.mp3", test_samples);
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g_list_free (test_samples);
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return 0;
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}
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