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gst/audiotestsrc/gstaudiotestsrc.*: Adjust to some recent api changes and add wtays new cool seeking capabillities
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
parent
534e0c268e
commit
7906b3c945
3 changed files with 100 additions and 83 deletions
11
ChangeLog
11
ChangeLog
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@ -1,3 +1,14 @@
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2005-12-14 Stefan Kost <ensonic@users.sf.net>
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* gst/audiotestsrc/gstaudiotestsrc.c:
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(gst_audio_test_src_class_init), (gst_audio_test_src_init),
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(gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
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(gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
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(gst_audio_test_src_create):
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* gst/audiotestsrc/gstaudiotestsrc.h:
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Adjust to some recent api changes and add wtays new cool seeking
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capabillities
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2005-12-14 Tim-Philipp Müller <tim at centricular dot net>
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* ext/alsa/Makefile.am:
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@ -121,8 +121,11 @@ static gboolean gst_audio_test_src_setcaps (GstBaseSrc * basesrc,
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GstCaps * caps);
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static void gst_audio_test_src_src_fixate (GstPad * pad, GstCaps * caps);
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static const GstQueryType *gst_audio_test_src_get_query_types (GstPad * pad);
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static gboolean gst_audio_test_src_src_query (GstPad * pad, GstQuery * query);
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static gboolean gst_audio_test_src_is_seekable (GstBaseSrc * basesrc);
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static gboolean gst_audio_test_src_do_seek (GstBaseSrc * basesrc,
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GstSegment * segment);
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static gboolean gst_audio_test_src_src_query (GstBaseSrc * basesrc,
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GstQuery * query);
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static void gst_audio_test_src_change_wave (GstAudioTestSrc * src);
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@ -130,7 +133,6 @@ static void gst_audio_test_src_get_times (GstBaseSrc * basesrc,
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GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
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static GstFlowReturn gst_audio_test_src_create (GstBaseSrc * basesrc,
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guint64 offset, guint length, GstBuffer ** buffer);
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static gboolean gst_audio_test_src_start (GstBaseSrc * basesrc);
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static void
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@ -178,7 +180,10 @@ gst_audio_test_src_class_init (GstAudioTestSrcClass * klass)
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G_MAXINT64, 0, G_PARAM_READWRITE));
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gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_test_src_setcaps);
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gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_audio_test_src_start);
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gstbasesrc_class->is_seekable =
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GST_DEBUG_FUNCPTR (gst_audio_test_src_is_seekable);
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gstbasesrc_class->do_seek = GST_DEBUG_FUNCPTR (gst_audio_test_src_do_seek);
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gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_audio_test_src_src_query);
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gstbasesrc_class->get_times =
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GST_DEBUG_FUNCPTR (gst_audio_test_src_get_times);
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gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_audio_test_src_create);
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@ -190,17 +195,15 @@ gst_audio_test_src_init (GstAudioTestSrc * src, GstAudioTestSrcClass * g_class)
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GstPad *pad = GST_BASE_SRC_PAD (src);
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gst_pad_set_fixatecaps_function (pad, gst_audio_test_src_src_fixate);
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gst_pad_set_query_function (pad, gst_audio_test_src_src_query);
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gst_pad_set_query_type_function (pad, gst_audio_test_src_get_query_types);
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src->samplerate = 44100;
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src->volume = 1.0;
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src->freq = 440.0;
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/* we operate in time */
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gst_base_src_set_format (GST_BASE_SRC (src), GST_FORMAT_TIME);
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gst_base_src_set_live (GST_BASE_SRC (src), FALSE);
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src->samples_per_buffer = 1024;
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src->timestamp = G_GINT64_CONSTANT (0);
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src->offset = G_GINT64_CONSTANT (0);
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src->timestamp_offset = G_GINT64_CONSTANT (0);
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src->wave = GST_AUDIO_TEST_SRC_WAVE_SINE;
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@ -220,73 +223,61 @@ gst_audio_test_src_src_fixate (GstPad * pad, GstCaps * caps)
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static gboolean
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gst_audio_test_src_setcaps (GstBaseSrc * basesrc, GstCaps * caps)
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{
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GstAudioTestSrc *audiotestsrc;
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GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (basesrc);
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const GstStructure *structure;
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gboolean ret;
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audiotestsrc = GST_AUDIO_TEST_SRC (basesrc);
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structure = gst_caps_get_structure (caps, 0);
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ret = gst_structure_get_int (structure, "rate", &audiotestsrc->samplerate);
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ret = gst_structure_get_int (structure, "rate", &src->samplerate);
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return ret;
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}
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static const GstQueryType *
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gst_audio_test_src_get_query_types (GstPad * pad)
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{
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static const GstQueryType query_types[] = {
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GST_QUERY_POSITION,
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0,
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};
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return query_types;
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}
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static gboolean
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gst_audio_test_src_src_query (GstPad * pad, GstQuery * query)
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gst_audio_test_src_src_query (GstBaseSrc * basesrc, GstQuery * query)
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{
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GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (basesrc);
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gboolean res = FALSE;
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GstAudioTestSrc *src;
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src = GST_AUDIO_TEST_SRC (GST_PAD_PARENT (pad));
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_POSITION:
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case GST_QUERY_CONVERT:
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{
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GstFormat format;
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gint64 current;
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GstFormat src_fmt, dest_fmt;
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gint64 src_val, dest_val;
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gst_query_parse_position (query, &format, NULL);
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gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
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if (src_fmt == dest_fmt) {
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dest_val = src_val;
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goto done;
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}
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switch (format) {
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switch (src_fmt) {
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case GST_FORMAT_DEFAULT:
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switch (dest_fmt) {
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case GST_FORMAT_TIME:
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/* samples to time */
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dest_val = src_val / src->samplerate;
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break;
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default:
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goto error;
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}
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break;
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case GST_FORMAT_TIME:
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current = src->timestamp;
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res = TRUE;
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break;
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case GST_FORMAT_DEFAULT: /* samples */
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current = src->offset / 2; /* 16bpp audio */
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res = TRUE;
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break;
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case GST_FORMAT_BYTES:
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current = src->offset;
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res = TRUE;
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switch (dest_fmt) {
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case GST_FORMAT_DEFAULT:
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/* time to samples */
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dest_val = src_val * src->samplerate;
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break;
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default:
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goto error;
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}
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break;
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default:
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break;
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goto error;
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}
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if (res) {
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gst_query_set_position (query, format, current);
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}
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break;
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}
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case GST_QUERY_DURATION:
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{
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GstFormat format;
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/* unlimited length */
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gst_query_parse_duration (query, &format, NULL);
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gst_query_set_duration (query, format, -1);
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done:
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gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
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res = TRUE;
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break;
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}
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default:
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@ -294,6 +285,12 @@ gst_audio_test_src_src_query (GstPad * pad, GstQuery * query)
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}
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return res;
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/* ERROR */
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error:
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{
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GST_DEBUG_OBJECT (src, "query failed");
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return FALSE;
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}
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}
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static void
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@ -532,13 +529,37 @@ gst_audio_test_src_get_times (GstBaseSrc * basesrc, GstBuffer * buffer,
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}
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}
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static gboolean
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gst_audio_test_src_do_seek (GstBaseSrc * basesrc, GstSegment * segment)
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{
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GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (basesrc);
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GstClockTime time;
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time = segment->time = segment->start;
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/* now move to the time indicated */
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src->n_samples = time * src->samplerate / GST_SECOND;
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src->running_time = src->n_samples * GST_SECOND / src->samplerate;
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g_assert (src->running_time <= time);
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return TRUE;
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}
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static gboolean
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gst_audio_test_src_is_seekable (GstBaseSrc * basesrc)
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{
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/* we're seekable... */
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return TRUE;
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}
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static GstFlowReturn
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gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset,
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guint length, GstBuffer ** buffer)
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{
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GstAudioTestSrc *src;
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GstBuffer *buf;
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guint tdiff;
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GstClockTime next_time;
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src = GST_AUDIO_TEST_SRC (basesrc);
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@ -556,21 +577,20 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset,
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src->tags_pushed = TRUE;
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}
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tdiff = src->samples_per_buffer * GST_SECOND / src->samplerate;
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buf = gst_buffer_new_and_alloc (src->samples_per_buffer * sizeof (gint16));
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gst_buffer_set_caps (buf, GST_PAD_CAPS (basesrc->srcpad));
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GST_BUFFER_TIMESTAMP (buf) = src->timestamp + src->timestamp_offset;
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GST_BUFFER_TIMESTAMP (buf) = src->timestamp_offset + src->running_time;
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/* offset is the number of samples */
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GST_BUFFER_OFFSET (buf) = src->offset;
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GST_BUFFER_OFFSET_END (buf) = src->offset + src->samples_per_buffer;
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GST_BUFFER_DURATION (buf) = tdiff;
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GST_BUFFER_OFFSET (buf) = src->n_samples;
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src->n_samples += src->samples_per_buffer;
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GST_BUFFER_OFFSET_END (buf) = src->n_samples;
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next_time = src->n_samples * GST_SECOND / src->samplerate;
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GST_BUFFER_DURATION (buf) = next_time - src->running_time;
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gst_object_sync_values (G_OBJECT (src), src->timestamp);
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gst_object_sync_values (G_OBJECT (src), src->running_time);
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src->timestamp += tdiff;
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src->offset += src->samples_per_buffer;
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src->running_time = next_time;
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src->process (src, (gint16 *) GST_BUFFER_DATA (buf));
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@ -642,17 +662,6 @@ gst_audio_test_src_get_property (GObject * object, guint prop_id,
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}
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}
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static gboolean
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gst_audio_test_src_start (GstBaseSrc * basesrc)
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{
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GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (basesrc);
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src->timestamp = G_GINT64_CONSTANT (0);
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src->offset = G_GINT64_CONSTANT (0);
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return TRUE;
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}
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static gboolean
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plugin_init (GstPlugin * plugin)
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{
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@ -79,20 +79,17 @@ struct _GstAudioTestSrc {
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/* audio parameters */
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gint samplerate;
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gint samples_per_buffer;
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guint64 timestamp;
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guint64 offset;
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gdouble accumulator;
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gboolean tags_pushed;
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GstClockID clock_id;
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GstClockTimeDiff timestamp_offset;
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/* < private > */
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GstClockID clock_id;
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GstClockTimeDiff timestamp_offset; /* base offset */
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GstClockTime running_time; /* total running time */
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gint64 n_samples; /* total samples sent */
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/* waveform specific context data */
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GstPinkNoise pink;
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