--- a/gst_plugins_base/gst/audiotestsrc/gstaudiotestsrc.c Wed Mar 24 17:58:42 2010 -0500
+++ b/gst_plugins_base/gst/audiotestsrc/gstaudiotestsrc.c Wed Mar 24 18:04:17 2010 -0500
@@ -19,26 +19,20 @@
/**
* SECTION:element-audiotestsrc
*
+ * AudioTestSrc can be used to generate basic audio signals. It support several
+ * different waveforms and allows to set the base frequency and volume.
+ *
* <refsect2>
- * <para>
- * AudioTestSrc can be used to generate basic audio signals. It support several
- * different waveforms and allows you to set the base frequency and volume.
- * </para>
* <title>Example launch line</title>
- * <para>
- * <programlisting>
+ * |[
* gst-launch audiotestsrc ! audioconvert ! alsasink
- * </programlisting>
- * This pipeline produces a sine with default frequency (mid-C) and volume.
- * </para>
- * <para>
- * <programlisting>
+ * ]| This pipeline produces a sine with default frequency, 440 Hz, and the
+ * default volume, 0.8 (relative to a maximum 1.0).
+ * |[
* gst-launch audiotestsrc wave=2 freq=200 ! audioconvert ! tee name=t ! queue ! alsasink t. ! queue ! libvisual_lv_scope ! ffmpegcolorspace ! xvimagesink
- * </programlisting>
- * In this example a saw wave is generated. The wave is shown using a
+ * ]| In this example a saw wave is generated. The wave is shown using a
* scope visualizer from libvisual, allowing you to visually verify that
* the saw wave is correct.
- * </para>
* </refsect2>
*/
@@ -50,7 +44,7 @@
#include <stdlib.h>
#include <string.h>
#include <gst/controller/gstcontroller.h>
-#include <glib_global.h>
+
#include "gstaudiotestsrc.h"
@@ -73,6 +67,14 @@
"Creates audio test signals of given frequency and volume",
"Stefan Kost <ensonic@users.sf.net>");
+#define DEFAULT_SAMPLES_PER_BUFFER 1024
+#define DEFAULT_WAVE GST_AUDIO_TEST_SRC_WAVE_SINE
+#define DEFAULT_FREQ 440.0
+#define DEFAULT_VOLUME 0.8
+#define DEFAULT_IS_LIVE FALSE
+#define DEFAULT_TIMESTAMP_OFFSET G_GINT64_CONSTANT (0)
+#define DEFAULT_CAN_ACTIVATE_PUSH TRUE
+#define DEFAULT_CAN_ACTIVATE_PULL FALSE
enum
{
@@ -83,6 +85,9 @@
PROP_VOLUME,
PROP_IS_LIVE,
PROP_TIMESTAMP_OFFSET,
+ PROP_CAN_ACTIVATE_PUSH,
+ PROP_CAN_ACTIVATE_PULL,
+ PROP_LAST
};
@@ -96,18 +101,18 @@
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], "
- "channels = (int) 1; "
+ "channels = (int) [ 1, 2 ]; "
"audio/x-raw-int, "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) true, "
"width = (int) 32, "
"depth = (int) 32,"
"rate = (int) [ 1, MAX ], "
- "channels = (int) 1; "
+ "channels = (int) [ 1, 2 ]; "
"audio/x-raw-float, "
"endianness = (int) BYTE_ORDER, "
"width = (int) { 32, 64 }, "
- "rate = (int) [ 1, MAX ], " "channels = (int) 1")
+ "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
);
@@ -125,9 +130,12 @@
{GST_AUDIO_TEST_SRC_WAVE_SAW, "Saw", "saw"},
{GST_AUDIO_TEST_SRC_WAVE_TRIANGLE, "Triangle", "triangle"},
{GST_AUDIO_TEST_SRC_WAVE_SILENCE, "Silence", "silence"},
- {GST_AUDIO_TEST_SRC_WAVE_WHITE_NOISE, "White noise", "white-noise"},
+ {GST_AUDIO_TEST_SRC_WAVE_WHITE_NOISE, "White uniform 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"},
+ {GST_AUDIO_TEST_SRC_WAVE_SINE_TAB, "Sine table", "sine-table"},
+ {GST_AUDIO_TEST_SRC_WAVE_TICKS, "Periodic Ticks", "ticks"},
+ {GST_AUDIO_TEST_SRC_WAVE_GAUSSIAN_WHITE_NOISE, "White Gaussian noise",
+ "gaussian-noise"},
{0, NULL, NULL},
};
@@ -148,6 +156,7 @@
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_check_get_range (GstBaseSrc * basesrc);
static gboolean gst_audio_test_src_do_seek (GstBaseSrc * basesrc,
GstSegment * segment);
static gboolean gst_audio_test_src_query (GstBaseSrc * basesrc,
@@ -157,6 +166,8 @@
static void gst_audio_test_src_get_times (GstBaseSrc * basesrc,
GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
+static gboolean gst_audio_test_src_start (GstBaseSrc * basesrc);
+static gboolean gst_audio_test_src_stop (GstBaseSrc * basesrc);
static GstFlowReturn gst_audio_test_src_create (GstBaseSrc * basesrc,
guint64 offset, guint length, GstBuffer ** buffer);
@@ -186,32 +197,50 @@
g_object_class_install_property (gobject_class, PROP_SAMPLES_PER_BUFFER,
g_param_spec_int ("samplesperbuffer", "Samples per buffer",
"Number of samples in each outgoing buffer",
- 1, G_MAXINT, 1024, G_PARAM_READWRITE));
- g_object_class_install_property (gobject_class, PROP_WAVE, g_param_spec_enum ("wave", "Waveform", "Oscillator waveform", GST_TYPE_AUDIO_TEST_SRC_WAVE, /* enum type */
- GST_AUDIO_TEST_SRC_WAVE_SINE, /* default value */
- G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ 1, G_MAXINT, DEFAULT_SAMPLES_PER_BUFFER,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_WAVE,
+ g_param_spec_enum ("wave", "Waveform", "Oscillator waveform",
+ GST_TYPE_AUDIO_TEST_SRC_WAVE, GST_AUDIO_TEST_SRC_WAVE_SINE,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_FREQ,
g_param_spec_double ("freq", "Frequency", "Frequency of test signal",
- 0.0, 20000.0, 440.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ 0.0, 20000.0, DEFAULT_FREQ,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_VOLUME,
- g_param_spec_double ("volume", "Volume", "Volume of test signal",
- 0.0, 1.0, 0.8, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_param_spec_double ("volume", "Volume", "Volume of test signal", 0.0,
+ 1.0, DEFAULT_VOLUME,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_IS_LIVE,
g_param_spec_boolean ("is-live", "Is Live",
- "Whether to act as a live source", FALSE, G_PARAM_READWRITE));
+ "Whether to act as a live source", DEFAULT_IS_LIVE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
- PROP_TIMESTAMP_OFFSET,
- g_param_spec_int64 ("timestamp-offset", "Timestamp offset",
+ PROP_TIMESTAMP_OFFSET, g_param_spec_int64 ("timestamp-offset",
+ "Timestamp offset",
"An offset added to timestamps set on buffers (in ns)", G_MININT64,
- G_MAXINT64, 0, G_PARAM_READWRITE));
+ G_MAXINT64, DEFAULT_TIMESTAMP_OFFSET,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_CAN_ACTIVATE_PUSH,
+ g_param_spec_boolean ("can-activate-push", "Can activate push",
+ "Can activate in push mode", DEFAULT_CAN_ACTIVATE_PUSH,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_CAN_ACTIVATE_PULL,
+ g_param_spec_boolean ("can-activate-pull", "Can activate pull",
+ "Can activate in pull mode", DEFAULT_CAN_ACTIVATE_PULL,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_test_src_setcaps);
gstbasesrc_class->is_seekable =
GST_DEBUG_FUNCPTR (gst_audio_test_src_is_seekable);
+ gstbasesrc_class->check_get_range =
+ GST_DEBUG_FUNCPTR (gst_audio_test_src_check_get_range);
gstbasesrc_class->do_seek = GST_DEBUG_FUNCPTR (gst_audio_test_src_do_seek);
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->start = GST_DEBUG_FUNCPTR (gst_audio_test_src_start);
+ gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_audio_test_src_stop);
gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_audio_test_src_create);
}
@@ -224,18 +253,21 @@
src->samplerate = 44100;
src->format = GST_AUDIO_TEST_SRC_FORMAT_NONE;
- src->volume = 0.8;
- src->freq = 440.0;
+
+ src->volume = DEFAULT_VOLUME;
+ src->freq = DEFAULT_FREQ;
/* 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);
+ gst_base_src_set_live (GST_BASE_SRC (src), DEFAULT_IS_LIVE);
- src->samples_per_buffer = 1024;
+ src->samples_per_buffer = DEFAULT_SAMPLES_PER_BUFFER;
src->generate_samples_per_buffer = src->samples_per_buffer;
- src->timestamp_offset = G_GINT64_CONSTANT (0);
+ src->timestamp_offset = DEFAULT_TIMESTAMP_OFFSET;
+ src->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL;
- src->wave = GST_AUDIO_TEST_SRC_WAVE_SINE;
+ src->wave = DEFAULT_WAVE;
+ gst_base_src_set_blocksize (GST_BASE_SRC (src), -1);
}
static void
@@ -247,6 +279,8 @@
structure = gst_caps_get_structure (caps, 0);
+ GST_DEBUG_OBJECT (src, "fixating samplerate to %d", src->samplerate);
+
gst_structure_fixate_field_nearest_int (structure, "rate", src->samplerate);
name = gst_structure_get_name (structure);
@@ -254,6 +288,9 @@
gst_structure_fixate_field_nearest_int (structure, "width", 32);
else if (strcmp (name, "audio/x-raw-float") == 0)
gst_structure_fixate_field_nearest_int (structure, "width", 64);
+
+ /* fixate to mono unless downstream requires stereo, for backwards compat */
+ gst_structure_fixate_field_nearest_int (structure, "channels", 1);
}
static gboolean
@@ -268,6 +305,8 @@
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "rate", &src->samplerate);
+ GST_DEBUG_OBJECT (src, "negotiated to samplerate %d", src->samplerate);
+
name = gst_structure_get_name (structure);
if (strcmp (name, "audio/x-raw-int") == 0) {
ret &= gst_structure_get_int (structure, "width", &width);
@@ -279,6 +318,29 @@
GST_AUDIO_TEST_SRC_FORMAT_F64;
}
+ /* allocate a new buffer suitable for this pad */
+ switch (src->format) {
+ case GST_AUDIO_TEST_SRC_FORMAT_S16:
+ src->sample_size = sizeof (gint16);
+ break;
+ case GST_AUDIO_TEST_SRC_FORMAT_S32:
+ src->sample_size = sizeof (gint32);
+ break;
+ case GST_AUDIO_TEST_SRC_FORMAT_F32:
+ src->sample_size = sizeof (gfloat);
+ break;
+ case GST_AUDIO_TEST_SRC_FORMAT_F64:
+ src->sample_size = sizeof (gdouble);
+ break;
+ default:
+ /* can't really happen */
+ ret = FALSE;
+ break;
+ }
+
+ ret &= gst_structure_get_int (structure, "channels", &src->channels);
+ GST_DEBUG_OBJECT (src, "negotiated to %d channels", src->channels);
+
gst_audio_test_src_change_wave (src);
return ret;
@@ -353,18 +415,21 @@
static void \
gst_audio_test_src_create_sine_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
- gint i; \
+ gint i, c; \
gdouble step, amp; \
\
step = M_PI_M2 * src->freq / src->samplerate; \
amp = src->volume * scale; \
\
- for (i = 0; i < src->generate_samples_per_buffer; i++) { \
+ i = 0; \
+ while (i < (src->generate_samples_per_buffer * src->channels)) { \
src->accumulator += step; \
if (src->accumulator >= M_PI_M2) \
src->accumulator -= M_PI_M2; \
\
- samples[i] = (g##type) (sin (src->accumulator) * amp); \
+ for (c = 0; c < src->channels; ++c) { \
+ samples[i++] = (g##type) (sin (src->accumulator) * amp); \
+ } \
} \
}
@@ -373,7 +438,7 @@
DEFINE_SINE (float, 1.0);
DEFINE_SINE (double, 1.0);
-static ProcessFunc sine_funcs[] = {
+static const ProcessFunc sine_funcs[] = {
(ProcessFunc) gst_audio_test_src_create_sine_int16,
(ProcessFunc) gst_audio_test_src_create_sine_int32,
(ProcessFunc) gst_audio_test_src_create_sine_float,
@@ -384,18 +449,21 @@
static void \
gst_audio_test_src_create_square_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
- gint i; \
+ gint i, c; \
gdouble step, amp; \
\
step = M_PI_M2 * src->freq / src->samplerate; \
amp = src->volume * scale; \
\
- for (i = 0; i < src->generate_samples_per_buffer; i++) { \
+ i = 0; \
+ while (i < (src->generate_samples_per_buffer * src->channels)) { \
src->accumulator += step; \
if (src->accumulator >= M_PI_M2) \
src->accumulator -= M_PI_M2; \
\
- samples[i] = (g##type) ((src->accumulator < M_PI) ? amp : -amp); \
+ for (c = 0; c < src->channels; ++c) { \
+ samples[i++] = (g##type) ((src->accumulator < M_PI) ? amp : -amp); \
+ } \
} \
}
@@ -404,7 +472,7 @@
DEFINE_SQUARE (float, 1.0);
DEFINE_SQUARE (double, 1.0);
-static ProcessFunc square_funcs[] = {
+static const ProcessFunc square_funcs[] = {
(ProcessFunc) gst_audio_test_src_create_square_int16,
(ProcessFunc) gst_audio_test_src_create_square_int32,
(ProcessFunc) gst_audio_test_src_create_square_float,
@@ -415,21 +483,24 @@
static void \
gst_audio_test_src_create_saw_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
- gint i; \
+ gint i, c; \
gdouble step, amp; \
\
step = M_PI_M2 * src->freq / src->samplerate; \
amp = (src->volume * scale) / M_PI; \
\
- for (i = 0; i < src->generate_samples_per_buffer; i++) { \
+ i = 0; \
+ while (i < (src->generate_samples_per_buffer * src->channels)) { \
src->accumulator += step; \
if (src->accumulator >= M_PI_M2) \
src->accumulator -= M_PI_M2; \
\
if (src->accumulator < M_PI) { \
- samples[i] = (g##type) (src->accumulator * amp); \
+ for (c = 0; c < src->channels; ++c) \
+ samples[i++] = (g##type) (src->accumulator * amp); \
} else { \
- samples[i] = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
+ for (c = 0; c < src->channels; ++c) \
+ samples[i++] = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
} \
} \
}
@@ -439,7 +510,7 @@
DEFINE_SAW (float, 1.0);
DEFINE_SAW (double, 1.0);
-static ProcessFunc saw_funcs[] = {
+static const ProcessFunc saw_funcs[] = {
(ProcessFunc) gst_audio_test_src_create_saw_int16,
(ProcessFunc) gst_audio_test_src_create_saw_int32,
(ProcessFunc) gst_audio_test_src_create_saw_float,
@@ -450,23 +521,27 @@
static void \
gst_audio_test_src_create_triangle_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
- gint i; \
+ gint i, c; \
gdouble step, amp; \
\
step = M_PI_M2 * src->freq / src->samplerate; \
amp = (src->volume * scale) / M_PI_2; \
\
- for (i = 0; i < src->generate_samples_per_buffer; i++) { \
+ i = 0; \
+ while (i < (src->generate_samples_per_buffer * src->channels)) { \
src->accumulator += step; \
if (src->accumulator >= M_PI_M2) \
src->accumulator -= M_PI_M2; \
\
if (src->accumulator < (M_PI * 0.5)) { \
- samples[i] = (g##type) (src->accumulator * amp); \
+ for (c = 0; c < src->channels; ++c) \
+ samples[i++] = (g##type) (src->accumulator * amp); \
} else if (src->accumulator < (M_PI * 1.5)) { \
- samples[i] = (g##type) ((src->accumulator - M_PI) * -amp); \
+ for (c = 0; c < src->channels; ++c) \
+ samples[i++] = (g##type) ((src->accumulator - M_PI) * -amp); \
} else { \
- samples[i] = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
+ for (c = 0; c < src->channels; ++c) \
+ samples[i++] = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
} \
} \
}
@@ -476,7 +551,7 @@
DEFINE_TRIANGLE (float, 1.0);
DEFINE_TRIANGLE (double, 1.0);
-static ProcessFunc triangle_funcs[] = {
+static const ProcessFunc triangle_funcs[] = {
(ProcessFunc) gst_audio_test_src_create_triangle_int16,
(ProcessFunc) gst_audio_test_src_create_triangle_int32,
(ProcessFunc) gst_audio_test_src_create_triangle_float,
@@ -487,7 +562,7 @@
static void \
gst_audio_test_src_create_silence_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
- memset (samples, 0, src->generate_samples_per_buffer * sizeof (g##type)); \
+ memset (samples, 0, src->generate_samples_per_buffer * sizeof (g##type) * src->channels); \
}
DEFINE_SILENCE (int16);
@@ -495,7 +570,7 @@
DEFINE_SILENCE (float);
DEFINE_SILENCE (double);
-static ProcessFunc silence_funcs[] = {
+static const ProcessFunc silence_funcs[] = {
(ProcessFunc) gst_audio_test_src_create_silence_int16,
(ProcessFunc) gst_audio_test_src_create_silence_int32,
(ProcessFunc) gst_audio_test_src_create_silence_float,
@@ -506,11 +581,13 @@
static void \
gst_audio_test_src_create_white_noise_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
- gint i; \
+ gint i, c; \
gdouble amp = (src->volume * scale); \
\
- for (i = 0; i < src->generate_samples_per_buffer; i++) { \
- samples[i] = (g##type) (amp * g_random_double_range (-1.0, 1.0)); \
+ i = 0; \
+ while (i < (src->generate_samples_per_buffer * src->channels)) { \
+ for (c = 0; c < src->channels; ++c) \
+ samples[i++] = (g##type) (amp * g_random_double_range (-1.0, 1.0)); \
} \
}
@@ -519,7 +596,7 @@
DEFINE_WHITE_NOISE (float, 1.0);
DEFINE_WHITE_NOISE (double, 1.0);
-static ProcessFunc white_noise_funcs[] = {
+static const ProcessFunc white_noise_funcs[] = {
(ProcessFunc) gst_audio_test_src_create_white_noise_int16,
(ProcessFunc) gst_audio_test_src_create_white_noise_int32,
(ProcessFunc) gst_audio_test_src_create_white_noise_float,
@@ -577,7 +654,6 @@
* values together. Only one changes each time.
*/
pink->running_sum -= pink->rows[num_zeros];
- //new_random = ((glong)GenerateRandomNumber()) >> PINK_RANDOM_SHIFT;
new_random = 32768.0 - (65536.0 * (gulong) rand () / (RAND_MAX + 1.0));
pink->running_sum += new_random;
pink->rows[num_zeros] = new_random;
@@ -595,15 +671,18 @@
static void \
gst_audio_test_src_create_pink_noise_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
- gint i; \
+ gint i, c; \
gdouble amp; \
\
amp = src->volume * scale; \
\
- for (i = 0; i < src->generate_samples_per_buffer; i++) { \
- samples[i] = \
+ i = 0; \
+ while (i < (src->generate_samples_per_buffer * src->channels)) { \
+ for (c = 0; c < src->channels; ++c) { \
+ samples[i++] = \
(g##type) (gst_audio_test_src_generate_pink_noise_value (&src->pink) * \
amp); \
+ } \
} \
}
@@ -612,7 +691,7 @@
DEFINE_PINK (float, 1.0);
DEFINE_PINK (double, 1.0);
-static ProcessFunc pink_noise_funcs[] = {
+static const ProcessFunc pink_noise_funcs[] = {
(ProcessFunc) gst_audio_test_src_create_pink_noise_int16,
(ProcessFunc) gst_audio_test_src_create_pink_noise_int32,
(ProcessFunc) gst_audio_test_src_create_pink_noise_float,
@@ -637,7 +716,40 @@
static void \
gst_audio_test_src_create_sine_table_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
- gint i; \
+ gint i, c; \
+ gdouble step, scl; \
+ \
+ step = M_PI_M2 * src->freq / src->samplerate; \
+ scl = 1024.0 / M_PI_M2; \
+ \
+ i = 0; \
+ while (i < (src->generate_samples_per_buffer * src->channels)) { \
+ src->accumulator += step; \
+ if (src->accumulator >= M_PI_M2) \
+ src->accumulator -= M_PI_M2; \
+ \
+ for (c = 0; c < src->channels; ++c) \
+ samples[i++] = (g##type) scale * src->wave_table[(gint) (src->accumulator * scl)]; \
+ } \
+}
+
+DEFINE_SINE_TABLE (int16, 32767.0);
+DEFINE_SINE_TABLE (int32, 2147483647.0);
+DEFINE_SINE_TABLE (float, 1.0);
+DEFINE_SINE_TABLE (double, 1.0);
+
+static const ProcessFunc sine_table_funcs[] = {
+ (ProcessFunc) gst_audio_test_src_create_sine_table_int16,
+ (ProcessFunc) gst_audio_test_src_create_sine_table_int32,
+ (ProcessFunc) gst_audio_test_src_create_sine_table_float,
+ (ProcessFunc) gst_audio_test_src_create_sine_table_double
+};
+
+#define DEFINE_TICKS(type,scale) \
+static void \
+gst_audio_test_src_create_tick_##type (GstAudioTestSrc * src, g##type * samples) \
+{ \
+ gint i, c; \
gdouble step, scl; \
\
step = M_PI_M2 * src->freq / src->samplerate; \
@@ -648,20 +760,64 @@
if (src->accumulator >= M_PI_M2) \
src->accumulator -= M_PI_M2; \
\
- samples[i] = (g##type) scale * src->wave_table[(gint) (src->accumulator * scl)]; \
+ if ((src->next_sample + i)%src->samplerate < 1600) { \
+ for (c = 0; c < src->channels; ++c) \
+ samples[(i * src->channels) + c] = (g##type) scale * src->wave_table[(gint) (src->accumulator * scl)]; \
+ } else { \
+ for (c = 0; c < src->channels; ++c) \
+ samples[(i * src->channels) + c] = 0; \
+ } \
} \
}
-DEFINE_SINE_TABLE (int16, 32767.0);
-DEFINE_SINE_TABLE (int32, 2147483647.0);
-DEFINE_SINE_TABLE (float, 1.0);
-DEFINE_SINE_TABLE (double, 1.0);
+DEFINE_TICKS (int16, 32767.0);
+DEFINE_TICKS (int32, 2147483647.0);
+DEFINE_TICKS (float, 1.0);
+DEFINE_TICKS (double, 1.0);
+
+static const ProcessFunc tick_funcs[] = {
+ (ProcessFunc) gst_audio_test_src_create_tick_int16,
+ (ProcessFunc) gst_audio_test_src_create_tick_int32,
+ (ProcessFunc) gst_audio_test_src_create_tick_float,
+ (ProcessFunc) gst_audio_test_src_create_tick_double
+};
+
+/* Gaussian white noise using Box-Muller algorithm. unit variance
+ * normally-distributed random numbers are generated in pairs as the real
+ * and imaginary parts of a compex random variable with
+ * uniformly-distributed argument and \chi^{2}-distributed modulus.
+ */
-static ProcessFunc sine_table_funcs[] = {
- (ProcessFunc) gst_audio_test_src_create_sine_table_int16,
- (ProcessFunc) gst_audio_test_src_create_sine_table_int32,
- (ProcessFunc) gst_audio_test_src_create_sine_table_float,
- (ProcessFunc) gst_audio_test_src_create_sine_table_double
+#define DEFINE_GAUSSIAN_WHITE_NOISE(type,scale) \
+static void \
+gst_audio_test_src_create_gaussian_white_noise_##type (GstAudioTestSrc * src, g##type * samples) \
+{ \
+ gint i, c; \
+ gdouble amp = (src->volume * scale); \
+ \
+ for (i = 0; i < src->generate_samples_per_buffer * src->channels; ) { \
+ for (c = 0; c < src->channels; ++c) { \
+ gdouble mag = sqrt (-2 * log (1.0 - g_random_double ())); \
+ gdouble phs = g_random_double_range (0.0, M_PI_M2); \
+ \
+ samples[i++] = (g##type) (amp * mag * cos (phs)); \
+ if (++c >= src->channels) \
+ break; \
+ samples[i++] = (g##type) (amp * mag * sin (phs)); \
+ } \
+ } \
+}
+
+DEFINE_GAUSSIAN_WHITE_NOISE (int16, 32767.0);
+DEFINE_GAUSSIAN_WHITE_NOISE (int32, 2147483647.0);
+DEFINE_GAUSSIAN_WHITE_NOISE (float, 1.0);
+DEFINE_GAUSSIAN_WHITE_NOISE (double, 1.0);
+
+static const ProcessFunc gaussian_white_noise_funcs[] = {
+ (ProcessFunc) gst_audio_test_src_create_gaussian_white_noise_int16,
+ (ProcessFunc) gst_audio_test_src_create_gaussian_white_noise_int32,
+ (ProcessFunc) gst_audio_test_src_create_gaussian_white_noise_float,
+ (ProcessFunc) gst_audio_test_src_create_gaussian_white_noise_double
};
/*
@@ -703,6 +859,13 @@
gst_audio_test_src_init_sine_table (src);
src->process = sine_table_funcs[src->format];
break;
+ case GST_AUDIO_TEST_SRC_WAVE_TICKS:
+ gst_audio_test_src_init_sine_table (src);
+ src->process = tick_funcs[src->format];
+ break;
+ case GST_AUDIO_TEST_SRC_WAVE_GAUSSIAN_WHITE_NOISE:
+ src->process = gaussian_white_noise_funcs[src->format];
+ break;
default:
GST_ERROR ("invalid wave-form");
break;
@@ -749,6 +912,30 @@
}
static gboolean
+gst_audio_test_src_start (GstBaseSrc * basesrc)
+{
+ GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (basesrc);
+
+ src->next_sample = 0;
+ src->next_byte = 0;
+ src->next_time = 0;
+ src->check_seek_stop = FALSE;
+ src->eos_reached = FALSE;
+ src->tags_pushed = FALSE;
+ src->accumulator = 0;
+
+ return TRUE;
+}
+
+static gboolean
+gst_audio_test_src_stop (GstBaseSrc * basesrc)
+{
+ return TRUE;
+}
+
+/* seek to time, will be called when we operate in push mode. In pull mode we
+ * get the requested byte offset. */
+static gboolean
gst_audio_test_src_do_seek (GstBaseSrc * basesrc, GstSegment * segment)
{
GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (basesrc);
@@ -758,16 +945,17 @@
time = segment->last_stop;
/* now move to the time indicated */
- src->n_samples =
+ src->next_sample =
gst_util_uint64_scale_int (time, src->samplerate, GST_SECOND);
- src->running_time =
- gst_util_uint64_scale_int (src->n_samples, GST_SECOND, src->samplerate);
+ src->next_byte = src->next_sample * src->sample_size * src->channels;
+ src->next_time =
+ gst_util_uint64_scale_int (src->next_sample, GST_SECOND, src->samplerate);
- g_assert (src->running_time <= time);
+ g_assert (src->next_time <= time);
if (GST_CLOCK_TIME_IS_VALID (segment->stop)) {
time = segment->stop;
- src->n_samples_stop = gst_util_uint64_scale_int (time, src->samplerate,
+ src->sample_stop = gst_util_uint64_scale_int (time, src->samplerate,
GST_SECOND);
src->check_seek_stop = TRUE;
} else {
@@ -785,6 +973,17 @@
return TRUE;
}
+static gboolean
+gst_audio_test_src_check_get_range (GstBaseSrc * basesrc)
+{
+ GstAudioTestSrc *src;
+
+ src = GST_AUDIO_TEST_SRC (basesrc);
+
+ /* if we can operate in pull mode */
+ return src->can_activate_pull;
+}
+
static GstFlowReturn
gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer ** buffer)
@@ -793,85 +992,97 @@
GstAudioTestSrc *src;
GstBuffer *buf;
GstClockTime next_time;
- gint64 n_samples;
- gint sample_size;
+ gint64 next_sample, next_byte;
+ guint bytes, samples;
+ GstElementClass *eclass;
src = GST_AUDIO_TEST_SRC (basesrc);
- if (src->eos_reached)
- return GST_FLOW_UNEXPECTED;
-
/* example for tagging generated data */
if (!src->tags_pushed) {
GstTagList *taglist;
- GstEvent *event;
taglist = gst_tag_list_new ();
gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
GST_TAG_DESCRIPTION, "audiotest wave", NULL);
- event = gst_event_new_tag (taglist);
- gst_pad_push_event (basesrc->srcpad, event);
+ eclass = GST_ELEMENT_CLASS (parent_class);
+ if (eclass->send_event)
+ eclass->send_event (GST_ELEMENT_CAST (basesrc),
+ gst_event_new_tag (taglist));
src->tags_pushed = TRUE;
}
+ if (src->eos_reached)
+ return GST_FLOW_UNEXPECTED;
+
+ /* if no length was given, use our default length in samples otherwise convert
+ * the length in bytes to samples. */
+ if (length == -1)
+ samples = src->samples_per_buffer;
+ else
+ samples = length / (src->sample_size * src->channels);
+
+ /* if no offset was given, use our next logical byte */
+ if (offset == -1)
+ offset = src->next_byte;
+
+ /* now see if we are at the byteoffset we think we are */
+ if (offset != src->next_byte) {
+ GST_DEBUG_OBJECT (src, "seek to new offset %" G_GUINT64_FORMAT, offset);
+ /* we have a discont in the expected sample offset, do a 'seek' */
+ src->next_sample = offset / (src->sample_size * src->channels);
+ src->next_time =
+ gst_util_uint64_scale_int (src->next_sample, GST_SECOND,
+ src->samplerate);
+ src->next_byte = offset;
+ }
+
/* check for eos */
if (src->check_seek_stop &&
- (src->n_samples_stop > src->n_samples) &&
- (src->n_samples_stop < src->n_samples + src->samples_per_buffer)
+ (src->sample_stop > src->next_sample) &&
+ (src->sample_stop < src->next_sample + samples)
) {
/* calculate only partial buffer */
- src->generate_samples_per_buffer = src->n_samples_stop - src->n_samples;
- n_samples = src->n_samples_stop;
+ src->generate_samples_per_buffer = src->sample_stop - src->next_sample;
+ next_sample = src->sample_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 = gst_util_uint64_scale (n_samples, GST_SECOND,
- (guint64) src->samplerate);
-
- /* allocate a new buffer suitable for this pad */
- switch (src->format) {
- case GST_AUDIO_TEST_SRC_FORMAT_S16:
- sample_size = sizeof (gint16);
- break;
- case GST_AUDIO_TEST_SRC_FORMAT_S32:
- sample_size = sizeof (gint32);
- break;
- case GST_AUDIO_TEST_SRC_FORMAT_F32:
- sample_size = sizeof (gfloat);
- break;
- case GST_AUDIO_TEST_SRC_FORMAT_F64:
- sample_size = sizeof (gdouble);
- break;
- default:
- sample_size = -1;
- GST_ELEMENT_ERROR (src, CORE, NEGOTIATION, (NULL),
- ("format wasn't negotiated before get function"));
- return GST_FLOW_NOT_NEGOTIATED;
- break;
+ src->generate_samples_per_buffer = samples;
+ next_sample = src->next_sample + samples;
}
- if ((res = gst_pad_alloc_buffer (basesrc->srcpad, src->n_samples,
- src->generate_samples_per_buffer * sample_size,
- GST_PAD_CAPS (basesrc->srcpad), &buf)) != GST_FLOW_OK) {
+ bytes = src->generate_samples_per_buffer * src->sample_size * src->channels;
+
+ if ((res = gst_pad_alloc_buffer (basesrc->srcpad, src->next_sample,
+ bytes, GST_PAD_CAPS (basesrc->srcpad), &buf)) != GST_FLOW_OK) {
return res;
}
- GST_BUFFER_TIMESTAMP (buf) = src->timestamp_offset + src->running_time;
- GST_BUFFER_OFFSET_END (buf) = n_samples;
- GST_BUFFER_DURATION (buf) = next_time - src->running_time;
+ next_byte = src->next_byte + bytes;
+ next_time = gst_util_uint64_scale_int (next_sample, GST_SECOND,
+ src->samplerate);
+
+ GST_LOG_OBJECT (src, "samplerate %d", src->samplerate);
+ GST_LOG_OBJECT (src, "next_sample %" G_GINT64_FORMAT ", ts %" GST_TIME_FORMAT,
+ next_sample, GST_TIME_ARGS (next_time));
- gst_object_sync_values (G_OBJECT (src), src->running_time);
+ GST_BUFFER_TIMESTAMP (buf) = src->timestamp_offset + src->next_time;
+ GST_BUFFER_OFFSET (buf) = src->next_sample;
+ GST_BUFFER_OFFSET_END (buf) = next_sample;
+ GST_BUFFER_DURATION (buf) = next_time - src->next_time;
- src->running_time = next_time;
- src->n_samples = n_samples;
+ gst_object_sync_values (G_OBJECT (src), src->next_time);
+
+ src->next_time = next_time;
+ src->next_sample = next_sample;
+ src->next_byte = next_byte;
GST_LOG_OBJECT (src, "generating %u samples at ts %" GST_TIME_FORMAT,
- length, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
+ src->generate_samples_per_buffer,
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
src->process (src, GST_BUFFER_DATA (buf));
@@ -912,6 +1123,12 @@
case PROP_TIMESTAMP_OFFSET:
src->timestamp_offset = g_value_get_int64 (value);
break;
+ case PROP_CAN_ACTIVATE_PUSH:
+ GST_BASE_SRC (src)->can_activate_push = g_value_get_boolean (value);
+ break;
+ case PROP_CAN_ACTIVATE_PULL:
+ src->can_activate_pull = g_value_get_boolean (value);
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@@ -943,6 +1160,12 @@
case PROP_TIMESTAMP_OFFSET:
g_value_set_int64 (value, src->timestamp_offset);
break;
+ case PROP_CAN_ACTIVATE_PUSH:
+ g_value_set_boolean (value, GST_BASE_SRC (src)->can_activate_push);
+ break;
+ case PROP_CAN_ACTIVATE_PULL:
+ g_value_set_boolean (value, src->can_activate_pull);
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;