mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-23 16:50:47 +00:00
Code Issues 1 Projects Releases Wiki Activity

gst/audiotestsrc/gstaudiotestsrc.*: Adjust to some recent api changes and add wtays new cool seeking capabillities

Browse source 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_init),
(gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
(gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
(gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
Adjust to some recent api changes and add wtays new cool seeking
capabillities
This commit is contained in:
Stefan Kost 2005-12-14 20:42:11 +00:00
parent 534e0c268e
commit 7906b3c945
3 changed files with 100 additions and 83 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

11
ChangeLog
View file

@ -1,3 +1,14 @@
2005-12-14 Stefan Kost <ensonic@users.sf.net>
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_init),
(gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
(gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
(gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
Adjust to some recent api changes and add wtays new cool seeking
capabillities
2005-12-14 Tim-Philipp Müller <tim at centricular dot net>
* ext/alsa/Makefile.am:

161
gst/audiotestsrc/gstaudiotestsrc.c
View file

@ -121,8 +121,11 @@ static gboolean gst_audio_test_src_setcaps (GstBaseSrc * basesrc,
GstCaps * caps);
static void gst_audio_test_src_src_fixate (GstPad * pad, GstCaps * caps);
static const GstQueryType *gst_audio_test_src_get_query_types (GstPad * pad);
static gboolean gst_audio_test_src_src_query (GstPad * pad, GstQuery * query);
static gboolean gst_audio_test_src_is_seekable (GstBaseSrc * basesrc);
static gboolean gst_audio_test_src_do_seek (GstBaseSrc * basesrc,
GstSegment * segment);
static gboolean gst_audio_test_src_src_query (GstBaseSrc * basesrc,
GstQuery * query);
static void gst_audio_test_src_change_wave (GstAudioTestSrc * src);
@ -130,7 +133,6 @@ static void gst_audio_test_src_get_times (GstBaseSrc * basesrc,
GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
static GstFlowReturn gst_audio_test_src_create (GstBaseSrc * basesrc,
guint64 offset, guint length, GstBuffer ** buffer);
static gboolean gst_audio_test_src_start (GstBaseSrc * basesrc);
static void
@ -178,7 +180,10 @@ gst_audio_test_src_class_init (GstAudioTestSrcClass * klass)
G_MAXINT64, 0, G_PARAM_READWRITE));
gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_test_src_setcaps);
gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_audio_test_src_start);
gstbasesrc_class->is_seekable =
GST_DEBUG_FUNCPTR (gst_audio_test_src_is_seekable);
gstbasesrc_class->do_seek = GST_DEBUG_FUNCPTR (gst_audio_test_src_do_seek);
gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_audio_test_src_src_query);
gstbasesrc_class->get_times =
GST_DEBUG_FUNCPTR (gst_audio_test_src_get_times);
gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_audio_test_src_create);
@ -190,17 +195,15 @@ gst_audio_test_src_init (GstAudioTestSrc * src, GstAudioTestSrcClass * g_class)
GstPad *pad = GST_BASE_SRC_PAD (src);
gst_pad_set_fixatecaps_function (pad, gst_audio_test_src_src_fixate);
gst_pad_set_query_function (pad, gst_audio_test_src_src_query);
gst_pad_set_query_type_function (pad, gst_audio_test_src_get_query_types);
src->samplerate = 44100;
src->volume = 1.0;
src->freq = 440.0;
/* we operate in time */
gst_base_src_set_format (GST_BASE_SRC (src), GST_FORMAT_TIME);
gst_base_src_set_live (GST_BASE_SRC (src), FALSE);
src->samples_per_buffer = 1024;
src->timestamp = G_GINT64_CONSTANT (0);
src->offset = G_GINT64_CONSTANT (0);
src->timestamp_offset = G_GINT64_CONSTANT (0);
src->wave = GST_AUDIO_TEST_SRC_WAVE_SINE;
@ -220,73 +223,61 @@ gst_audio_test_src_src_fixate (GstPad * pad, GstCaps * caps)
static gboolean
gst_audio_test_src_setcaps (GstBaseSrc * basesrc, GstCaps * caps)
{
GstAudioTestSrc *audiotestsrc;
GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (basesrc);
const GstStructure *structure;
gboolean ret;
audiotestsrc = GST_AUDIO_TEST_SRC (basesrc);
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "rate", &audiotestsrc->samplerate);
ret = gst_structure_get_int (structure, "rate", &src->samplerate);
return ret;
}
static const GstQueryType *
gst_audio_test_src_get_query_types (GstPad * pad)
{
static const GstQueryType query_types[] = {
GST_QUERY_POSITION,
0,
};
return query_types;
}
static gboolean
gst_audio_test_src_src_query (GstPad * pad, GstQuery * query)
gst_audio_test_src_src_query (GstBaseSrc * basesrc, GstQuery * query)
{
GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (basesrc);
gboolean res = FALSE;
GstAudioTestSrc *src;
src = GST_AUDIO_TEST_SRC (GST_PAD_PARENT (pad));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
case GST_QUERY_CONVERT:
{
GstFormat format;
gint64 current;
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
gst_query_parse_position (query, &format, NULL);
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
if (src_fmt == dest_fmt) {
dest_val = src_val;
goto done;
}
switch (format) {
switch (src_fmt) {
case GST_FORMAT_DEFAULT:
switch (dest_fmt) {
case GST_FORMAT_TIME:
/* samples to time */
dest_val = src_val / src->samplerate;
break;
default:
goto error;
}
break;
case GST_FORMAT_TIME:
current = src->timestamp;
res = TRUE;
break;
case GST_FORMAT_DEFAULT: /* samples */
current = src->offset / 2; /* 16bpp audio */
res = TRUE;
break;
case GST_FORMAT_BYTES:
current = src->offset;
res = TRUE;
switch (dest_fmt) {
case GST_FORMAT_DEFAULT:
/* time to samples */
dest_val = src_val * src->samplerate;
break;
default:
goto error;
}
break;
default:
break;
goto error;
}
if (res) {
gst_query_set_position (query, format, current);
}
break;
}
case GST_QUERY_DURATION:
{
GstFormat format;
/* unlimited length */
gst_query_parse_duration (query, &format, NULL);
gst_query_set_duration (query, format, -1);
done:
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
res = TRUE;
break;
}
default:
@ -294,6 +285,12 @@ gst_audio_test_src_src_query (GstPad * pad, GstQuery * query)
}
return res;
/* ERROR */
error:
{
GST_DEBUG_OBJECT (src, "query failed");
return FALSE;
}
}
static void
@ -532,13 +529,37 @@ gst_audio_test_src_get_times (GstBaseSrc * basesrc, GstBuffer * buffer,
}
}
static gboolean
gst_audio_test_src_do_seek (GstBaseSrc * basesrc, GstSegment * segment)
{
GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (basesrc);
GstClockTime time;
time = segment->time = segment->start;
/* now move to the time indicated */
src->n_samples = time * src->samplerate / GST_SECOND;
src->running_time = src->n_samples * GST_SECOND / src->samplerate;
g_assert (src->running_time <= time);
return TRUE;
}
static gboolean
gst_audio_test_src_is_seekable (GstBaseSrc * basesrc)
{
/* we're seekable... */
return TRUE;
}
static GstFlowReturn
gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer ** buffer)
{
GstAudioTestSrc *src;
GstBuffer *buf;
guint tdiff;
GstClockTime next_time;
src = GST_AUDIO_TEST_SRC (basesrc);
@ -556,21 +577,20 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset,
src->tags_pushed = TRUE;
}
tdiff = src->samples_per_buffer * GST_SECOND / src->samplerate;
buf = gst_buffer_new_and_alloc (src->samples_per_buffer * sizeof (gint16));
gst_buffer_set_caps (buf, GST_PAD_CAPS (basesrc->srcpad));
GST_BUFFER_TIMESTAMP (buf) = src->timestamp + src->timestamp_offset;
GST_BUFFER_TIMESTAMP (buf) = src->timestamp_offset + src->running_time;
/* offset is the number of samples */
GST_BUFFER_OFFSET (buf) = src->offset;
GST_BUFFER_OFFSET_END (buf) = src->offset + src->samples_per_buffer;
GST_BUFFER_DURATION (buf) = tdiff;
GST_BUFFER_OFFSET (buf) = src->n_samples;
src->n_samples += src->samples_per_buffer;
GST_BUFFER_OFFSET_END (buf) = src->n_samples;
next_time = src->n_samples * GST_SECOND / src->samplerate;
GST_BUFFER_DURATION (buf) = next_time - src->running_time;
gst_object_sync_values (G_OBJECT (src), src->timestamp);
gst_object_sync_values (G_OBJECT (src), src->running_time);
src->timestamp += tdiff;
src->offset += src->samples_per_buffer;
src->running_time = next_time;
src->process (src, (gint16 *) GST_BUFFER_DATA (buf));
@ -642,17 +662,6 @@ gst_audio_test_src_get_property (GObject * object, guint prop_id,
}
}
static gboolean
gst_audio_test_src_start (GstBaseSrc * basesrc)
{
GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (basesrc);
src->timestamp = G_GINT64_CONSTANT (0);
src->offset = G_GINT64_CONSTANT (0);
return TRUE;
}
static gboolean
plugin_init (GstPlugin * plugin)
{

11
gst/audiotestsrc/gstaudiotestsrc.h
View file

@ -79,20 +79,17 @@ struct _GstAudioTestSrc {
/* audio parameters */
gint samplerate;
gint samples_per_buffer;
guint64 timestamp;
guint64 offset;
gdouble accumulator;
gboolean tags_pushed;
GstClockID clock_id;
GstClockTimeDiff timestamp_offset;
/* < private > */
GstClockID clock_id;
GstClockTimeDiff timestamp_offset; /* base offset */
GstClockTime running_time; /* total running time */
gint64 n_samples; /* total samples sent */
/* waveform specific context data */
GstPinkNoise pink;