Commit 0d85e2cc authored by Stefan Kost's avatar Stefan Kost

gst/audiotestsrc/gstaudiotestsrc.*: update to basesrc changes, implement...

gst/audiotestsrc/gstaudiotestsrc.*: update to basesrc changes, implement segmented seeking and eos handling, add a 's...

Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
(gst_audio_test_src_init), (gst_audio_test_src_src_fixate),
(gst_audio_test_src_query), (gst_audio_test_src_create_sine),
(gst_audio_test_src_create_square),
(gst_audio_test_src_create_saw),
(gst_audio_test_src_create_triangle),
(gst_audio_test_src_create_silence),
(gst_audio_test_src_create_white_noise),
(gst_audio_test_src_create_pink_noise),
(gst_audio_test_src_init_sine_table),
(gst_audio_test_src_create_sine_table),
(gst_audio_test_src_change_wave),
(gst_audio_test_src_change_volume), (gst_audio_test_src_do_seek),
(gst_audio_test_src_create), (gst_audio_test_src_set_property):
* gst/audiotestsrc/gstaudiotestsrc.h:
update to basesrc changes, implement segmented seeking and eos handling,
add a 'sine-tab' waveform for performance critical playback
parent cd55b742
2005-12-29 Stefan Kost <ensonic@users.sf.net>
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
(gst_audio_test_src_init), (gst_audio_test_src_src_fixate),
(gst_audio_test_src_query), (gst_audio_test_src_create_sine),
(gst_audio_test_src_create_square),
(gst_audio_test_src_create_saw),
(gst_audio_test_src_create_triangle),
(gst_audio_test_src_create_silence),
(gst_audio_test_src_create_white_noise),
(gst_audio_test_src_create_pink_noise),
(gst_audio_test_src_init_sine_table),
(gst_audio_test_src_create_sine_table),
(gst_audio_test_src_change_wave),
(gst_audio_test_src_change_volume), (gst_audio_test_src_do_seek),
(gst_audio_test_src_create), (gst_audio_test_src_set_property):
* gst/audiotestsrc/gstaudiotestsrc.h:
update to basesrc changes, implement segmented seeking and eos handling,
add a 'sine-tab' waveform for performance critical playback
2005-12-29 Tim-Philipp Müller <tim at centricular dot net>
* po/POTFILES.in:
......
......@@ -53,6 +53,7 @@
#include "gstaudiotestsrc.h"
#define M_PI_M2 ( M_PI + M_PI )
GstElementDetails gst_audio_test_src_details = {
"Audio test source",
......@@ -102,6 +103,7 @@ gst_audiostestsrc_wave_get_type (void)
{GST_AUDIO_TEST_SRC_WAVE_SILENCE, "Silence", "silence"},
{GST_AUDIO_TEST_SRC_WAVE_WHITE_NOISE, "White noise", "white-noise"},
{GST_AUDIO_TEST_SRC_WAVE_PINK_NOISE, "Pink noise", "pink-noise"},
{GST_AUDIO_TEST_SRC_WAVE_SINE_TAB, "Sine table", "sine table"},
{0, NULL, NULL},
};
......@@ -124,7 +126,7 @@ static void gst_audio_test_src_src_fixate (GstPad * pad, GstCaps * caps);
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,
static gboolean gst_audio_test_src_query (GstBaseSrc * basesrc,
GstQuery * query);
static void gst_audio_test_src_change_wave (GstAudioTestSrc * src);
......@@ -183,7 +185,7 @@ gst_audio_test_src_class_init (GstAudioTestSrcClass * klass)
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->query = GST_DEBUG_FUNCPTR (gst_audio_test_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);
......@@ -204,6 +206,7 @@ gst_audio_test_src_init (GstAudioTestSrc * src, GstAudioTestSrcClass * g_class)
gst_base_src_set_live (GST_BASE_SRC (src), FALSE);
src->samples_per_buffer = 1024;
src->generate_samples_per_buffer = src->samples_per_buffer;
src->timestamp_offset = G_GINT64_CONSTANT (0);
src->wave = GST_AUDIO_TEST_SRC_WAVE_SINE;
......@@ -213,11 +216,12 @@ gst_audio_test_src_init (GstAudioTestSrc * src, GstAudioTestSrcClass * g_class)
static void
gst_audio_test_src_src_fixate (GstPad * pad, GstCaps * caps)
{
GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (GST_PAD_PARENT (pad));
GstStructure *structure;
structure = gst_caps_get_structure (caps, 0);
gst_structure_fixate_field_nearest_int (structure, "rate", 44100);
gst_structure_fixate_field_nearest_int (structure, "rate", src->samplerate);
}
static gboolean
......@@ -234,7 +238,7 @@ gst_audio_test_src_setcaps (GstBaseSrc * basesrc, GstCaps * caps)
}
static gboolean
gst_audio_test_src_src_query (GstBaseSrc * basesrc, GstQuery * query)
gst_audio_test_src_query (GstBaseSrc * basesrc, GstQuery * query)
{
GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (basesrc);
gboolean res = FALSE;
......@@ -299,13 +303,13 @@ gst_audio_test_src_create_sine (GstAudioTestSrc * src, gint16 * samples)
gint i;
gdouble step, amp;
step = 2 * M_PI * src->freq / src->samplerate;
step = M_PI_M2 * src->freq / src->samplerate;
amp = src->volume * 32767.0;
for (i = 0; i < src->samples_per_buffer; i++) {
for (i = 0; i < src->generate_samples_per_buffer; i++) {
src->accumulator += step;
if (src->accumulator >= 2 * M_PI)
src->accumulator -= 2 * M_PI;
if (src->accumulator >= M_PI_M2)
src->accumulator -= M_PI_M2;
samples[i] = (gint16) (sin (src->accumulator) * amp);
}
......@@ -317,13 +321,13 @@ gst_audio_test_src_create_square (GstAudioTestSrc * src, gint16 * samples)
gint i;
gdouble step, amp;
step = 2 * M_PI * src->freq / src->samplerate;
step = M_PI_M2 * src->freq / src->samplerate;
amp = src->volume * 32767.0;
for (i = 0; i < src->samples_per_buffer; i++) {
for (i = 0; i < src->generate_samples_per_buffer; i++) {
src->accumulator += step;
if (src->accumulator >= 2 * M_PI)
src->accumulator -= 2 * M_PI;
if (src->accumulator >= M_PI_M2)
src->accumulator -= M_PI_M2;
samples[i] = (gint16) ((src->accumulator < M_PI) ? amp : -amp);
}
......@@ -335,18 +339,18 @@ gst_audio_test_src_create_saw (GstAudioTestSrc * src, gint16 * samples)
gint i;
gdouble step, amp;
step = 2 * M_PI * src->freq / src->samplerate;
step = M_PI_M2 * src->freq / src->samplerate;
amp = (src->volume * 32767.0) / M_PI;
for (i = 0; i < src->samples_per_buffer; i++) {
for (i = 0; i < src->generate_samples_per_buffer; i++) {
src->accumulator += step;
if (src->accumulator >= 2 * M_PI)
src->accumulator -= 2 * M_PI;
if (src->accumulator >= M_PI_M2)
src->accumulator -= M_PI_M2;
if (src->accumulator < M_PI) {
samples[i] = (gint16) (src->accumulator * amp);
} else {
samples[i] = (gint16) ((2 * M_PI - src->accumulator) * -amp);
samples[i] = (gint16) ((M_PI_M2 - src->accumulator) * -amp);
}
}
}
......@@ -357,20 +361,20 @@ gst_audio_test_src_create_triangle (GstAudioTestSrc * src, gint16 * samples)
gint i;
gdouble step, amp;
step = 2 * M_PI * src->freq / src->samplerate;
amp = (src->volume * 32767.0) / (M_PI * 0.5);
step = M_PI_M2 * src->freq / src->samplerate;
amp = (src->volume * 32767.0) / M_PI_2;
for (i = 0; i < src->samples_per_buffer; i++) {
for (i = 0; i < src->generate_samples_per_buffer; i++) {
src->accumulator += step;
if (src->accumulator >= 2 * M_PI)
src->accumulator -= 2 * M_PI;
if (src->accumulator >= M_PI_M2)
src->accumulator -= M_PI_M2;
if (src->accumulator < (M_PI * 0.5)) {
samples[i] = (gint16) (src->accumulator * amp);
} else if (src->accumulator < (M_PI * 1.5)) {
samples[i] = (gint16) ((src->accumulator - M_PI) * -amp);
} else {
samples[i] = (gint16) ((2 * M_PI - src->accumulator) * -amp);
samples[i] = (gint16) ((M_PI_M2 - src->accumulator) * -amp);
}
}
}
......@@ -378,7 +382,7 @@ gst_audio_test_src_create_triangle (GstAudioTestSrc * src, gint16 * samples)
static void
gst_audio_test_src_create_silence (GstAudioTestSrc * src, gint16 * samples)
{
memset (samples, 0, src->samples_per_buffer * sizeof (gint16));
memset (samples, 0, src->generate_samples_per_buffer * sizeof (gint16));
}
static void
......@@ -389,7 +393,7 @@ gst_audio_test_src_create_white_noise (GstAudioTestSrc * src, gint16 * samples)
amp = src->volume * 65535.0;
for (i = 0; i < src->samples_per_buffer; i++) {
for (i = 0; i < src->generate_samples_per_buffer; i++) {
samples[i] = (gint16) (32768 - (amp * rand () / (RAND_MAX + 1.0)));
}
}
......@@ -467,13 +471,49 @@ gst_audio_test_src_create_pink_noise (GstAudioTestSrc * src, gint16 * samples)
amp = src->volume * 32767.0;
for (i = 0; i < src->samples_per_buffer; i++) {
for (i = 0; i < src->generate_samples_per_buffer; i++) {
samples[i] =
(gint16) (gst_audio_test_src_generate_pink_noise_value (&src->pink) *
amp);
}
}
static void
gst_audio_test_src_init_sine_table (GstAudioTestSrc * src)
{
gint i;
gdouble ang = 0.0;
gdouble step = M_PI_M2 / 1024.0;
gdouble amp = src->volume * 32767.0;
for (i = 0; i < 1024; i++) {
src->wave_table[i] = (gint16) (sin (ang) * amp);
ang += step;
}
}
static void
gst_audio_test_src_create_sine_table (GstAudioTestSrc * src, gint16 * samples)
{
gint i;
gdouble step, scl;
step = M_PI_M2 * src->freq / src->samplerate;
scl = 1024.0 / M_PI_M2;
for (i = 0; i < src->generate_samples_per_buffer; i++) {
src->accumulator += step;
if (src->accumulator >= M_PI_M2)
src->accumulator -= M_PI_M2;
samples[i] = (gint16) src->wave_table[(gint) (src->accumulator * scl)];
}
}
/*
* gst_audio_test_src_change_wave:
* Assign function pointer of wave genrator.
*/
static void
gst_audio_test_src_change_wave (GstAudioTestSrc * src)
{
......@@ -500,12 +540,32 @@ gst_audio_test_src_change_wave (GstAudioTestSrc * src)
gst_audio_test_src_init_pink_noise (src);
src->process = gst_audio_test_src_create_pink_noise;
break;
case GST_AUDIO_TEST_SRC_WAVE_SINE_TAB:
gst_audio_test_src_init_sine_table (src);
src->process = gst_audio_test_src_create_sine_table;
break;
default:
GST_ERROR ("invalid wave-form");
break;
}
}
/*
* gst_audio_test_src_change_volume:
* Recalc wave tables for precalculated waves.
*/
static void
gst_audio_test_src_change_volume (GstAudioTestSrc * src)
{
switch (src->wave) {
case GST_AUDIO_TEST_SRC_WAVE_SINE_TAB:
gst_audio_test_src_init_sine_table (src);
break;
default:
break;
}
}
static void
gst_audio_test_src_get_times (GstBaseSrc * basesrc, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
......@@ -543,17 +603,14 @@ gst_audio_test_src_do_seek (GstBaseSrc * basesrc, GstSegment * segment)
g_assert (src->running_time <= time);
/*
if (GST_CLOCK_TIME_IS_VALID (segment->stop)) {
time = segment->stop;
src->n_samples_stop = time * src->samplerate / GST_SECOND;
src->check_seek_stop = true;
src->seek_flags = segment.flags;
}
else {
src->check_seek_stop = false;
}
*/
if (GST_CLOCK_TIME_IS_VALID (segment->stop)) {
time = segment->stop;
src->n_samples_stop = time * src->samplerate / GST_SECOND;
src->check_seek_stop = TRUE;
} else {
src->check_seek_stop = FALSE;
}
src->eos_reached = FALSE;
return TRUE;
}
......@@ -572,9 +629,13 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset,
GstAudioTestSrc *src;
GstBuffer *buf;
GstClockTime next_time;
gint64 n_samples;
src = GST_AUDIO_TEST_SRC (basesrc);
if (src->eos_reached)
return GST_FLOW_UNEXPECTED;
if (!src->tags_pushed) {
GstTagList *taglist;
GstEvent *event;
......@@ -589,30 +650,35 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset,
src->tags_pushed = TRUE;
}
buf = gst_buffer_new_and_alloc (src->samples_per_buffer * sizeof (gint16));
if (src->check_seek_stop &&
(src->n_samples_stop > src->n_samples) &&
(src->n_samples_stop < src->n_samples + src->samples_per_buffer)
) {
/* calculate only partial buffer */
src->generate_samples_per_buffer = src->n_samples_stop - src->n_samples;
n_samples = src->n_samples_stop;
src->eos_reached = TRUE;
} else {
/* calculate full buffer */
src->generate_samples_per_buffer = src->samples_per_buffer;
n_samples = src->n_samples + src->samples_per_buffer;
}
next_time = n_samples * GST_SECOND / src->samplerate;
buf =
gst_buffer_new_and_alloc (src->generate_samples_per_buffer *
sizeof (gint16));
gst_buffer_set_caps (buf, GST_PAD_CAPS (basesrc->srcpad));
GST_BUFFER_TIMESTAMP (buf) = src->timestamp_offset + src->running_time;
/* offset is the number of samples */
GST_BUFFER_OFFSET (buf) = src->n_samples;
/*
if (src->check_seek_stop &&
(src->n_samples_stop > src->n_samples) &&
(src->n_samples_stop < src->n_samples + src->samples_per_buffer)) {
src->n_samples = src->n_samples_stop;
@todo: calculate only partial buffer!
@todo: send EOS or SEGMENT_DONE depending on segment.flags&GST_SEEK_FLAG_SEGMENT
}
else
*/
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_OFFSET_END (buf) = n_samples;
GST_BUFFER_DURATION (buf) = next_time - src->running_time;
gst_object_sync_values (G_OBJECT (src), src->running_time);
src->running_time = next_time;
src->n_samples = n_samples;
src->process (src, (gint16 *) GST_BUFFER_DATA (buf));
......@@ -640,6 +706,7 @@ gst_audio_test_src_set_property (GObject * object, guint prop_id,
break;
case PROP_VOLUME:
src->volume = g_value_get_double (value);
gst_audio_test_src_change_volume (src);
break;
case PROP_IS_LIVE:
gst_base_src_set_live (GST_BASE_SRC (src), g_value_get_boolean (value));
......
......@@ -50,6 +50,7 @@ typedef enum {
GST_AUDIO_TEST_SRC_WAVE_SILENCE,
GST_AUDIO_TEST_SRC_WAVE_WHITE_NOISE,
GST_AUDIO_TEST_SRC_WAVE_PINK_NOISE,
GST_AUDIO_TEST_SRC_WAVE_SINE_TAB
} GstAudioTestSrcWaves;
#define PINK_MAX_RANDOM_ROWS (30)
......@@ -81,23 +82,20 @@ struct _GstAudioTestSrc {
gint samplerate;
gint samples_per_buffer;
gdouble accumulator;
gboolean tags_pushed;
/* < private > */
GstClockID clock_id;
gboolean tags_pushed; /* send tags just once ? */
GstClockTimeDiff timestamp_offset; /* base offset */
GstClockTime running_time; /* total running time */
gint64 n_samples; /* total samples sent */
/*
gint64 n_samples_stop;
gboolean check_seek_stop;
GstSeekFlags seek_flags;
*/
gboolean eos_reached;
gint generate_samples_per_buffer; /* used to generate a partial buffer */
/* waveform specific context data */
gdouble accumulator; /* phase angle */
GstPinkNoise pink;
gint16 wave_table[1024];
};
struct _GstAudioTestSrcClass {
......
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