blob: 16e137d7577910aa98e4a18a6a67abee90931c9b [file] [log] [blame]
/* 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;
}