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gstreamer/gst/audiotestsrc/gstaudiotestsrc.c
Thibault Saunier 909baa2360 Pass the code through codespell
2019-08-30 13:05:36 +00:00

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/* GStreamer
* Copyright (C) 2005 Stefan Kost <ensonic@users.sf.net>
*
* 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.
*/
/**
* SECTION:element-audiotestsrc
* @title: audiotestsrc
*
* AudioTestSrc can be used to generate basic audio signals. It support several
* different waveforms and allows to set the base frequency and volume. Some
* waveforms might use additional properties.
*
* Waveform specific notes:
*
* ## Gaussian white noise
*
* This waveform produces white (zero mean) Gaussian noise.
* Volume sets the standard deviation of the noise in units of the range
* of values of the sample type, e.g. volume=0.1 produces noise with a
* standard deviation of 0.1*32767=3277 with 16-bit integer samples,
* or 0.1*1.0=0.1 with floating-point samples.
*
* ## Ticks
*
* This waveform is special in that it does not produce one continuous
* signal. Instead, it produces finite-length sine wave pulses (the "ticks").
* It is useful for detecting time shifts between audio signal, for example
* between RTSP audio clients that shall play synchronized. It is also useful
* for generating a signal that feeds the trigger of an oscilloscope.
*
* To further help with oscilloscope triggering and time offset detection,
* the waveform can apply a different volume to every Nth tick (this is then
* called the "marker tick"). For instance, one could generate a tick every
* 100ms, and make every 20th tick a marker tick (meaning that every 2 seconds
* there is a marker tick). This is useful for detecting large time offsets
* while still frequently triggering an oscilloscope.
*
* Also, a "ramp" can be applied to the begin & end of ticks. The sudden
* start of the sine tick is a discontinuity, even if the sine wave starts
* at 0. The* resulting artifacts can often make it more difficult to use the
* ticks for an oscilloscope's trigger. To that end, an initial "ramp" can
* be applied. The first few samples are modulated by a cubic function to
* reduce the impact of the discontinuity, resulting in smaller artifacts.
* The number of samples equals floor(samplerate / sine-wave-frequency).
* Example: with a sample rate of 48 kHz and a sine wave frequency of 10 kHz,
* the first 4 samples are modulated by the cubic function.
*
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc ! audioconvert ! autoaudiosink
* ]|
* This pipeline produces a sine with default frequency, 440 Hz, and the
* default volume, 0.8 (relative to a maximum 1.0).
* |[
* gst-launch-1.0 audiotestsrc wave=2 freq=200 ! tee name=t ! queue ! audioconvert ! \
* autoaudiosink t. ! queue ! audioconvert ! libvisual_lv_scope ! videoconvert ! autovideosink
* ]|
* 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.
*
* |[
* gst-launch-1.0 audiotestsrc wave=ticks apply-tick-ramp=true tick-interval=100000000 \
* freq=10000 volume=0.4 marker-tick-period=10 sine-periods-per-tick=20 ! autoaudiosink
* ]| This pipeline produces a series of 10 kHz sine wave ticks. Each tick is
* 20 sine wave periods long, ticks occur every 100 ms and have a volume of
* 0.4. Every 10th tick is a marker tick and has the default marker tick volume
* of 1.0. The beginning and end of the ticks are modulated with the ramp.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <math.h>
#include <stdlib.h>
#include <string.h>
#include "gstaudiotestsrc.h"
#define M_PI_M2 ( G_PI + G_PI )
GST_DEBUG_CATEGORY_STATIC (audio_test_src_debug);
#define GST_CAT_DEFAULT audio_test_src_debug
#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_SINE_PERIODS_PER_TICK 10
#define DEFAULT_TIME_BETWEEN_TICKS GST_SECOND
#define DEFAULT_MARKER_TICK_PERIOD 0
#define DEFAULT_MARKER_TICK_VOLUME 1.0
#define DEFAULT_APPLY_TICK_RAMP FALSE
#define DEFAULT_CAN_ACTIVATE_PUSH TRUE
#define DEFAULT_CAN_ACTIVATE_PULL FALSE
enum
{
PROP_0,
PROP_SAMPLES_PER_BUFFER,
PROP_WAVE,
PROP_FREQ,
PROP_VOLUME,
PROP_IS_LIVE,
PROP_TIMESTAMP_OFFSET,
PROP_SINE_PERIODS_PER_TICK,
PROP_TICK_INTERVAL,
PROP_MARKER_TICK_PERIOD,
PROP_MARKER_TICK_VOLUME,
PROP_APPLY_TICK_RAMP,
PROP_CAN_ACTIVATE_PUSH,
PROP_CAN_ACTIVATE_PULL
};
#define FORMAT_STR " { S16LE, S16BE, U16LE, U16BE, " \
"S24_32LE, S24_32BE, U24_32LE, U24_32BE, " \
"S32LE, S32BE, U32LE, U32BE, " \
"S24LE, S24BE, U24LE, U24BE, " \
"S20LE, S20BE, U20LE, U20BE, " \
"S18LE, S18BE, U18LE, U18BE, " \
"F32LE, F32BE, F64LE, F64BE, " \
"S8, U8 }"
#define DEFAULT_FORMAT_STR GST_AUDIO_NE ("S16")
static GstStaticPadTemplate gst_audio_test_src_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " FORMAT_STR ", "
"layout = (string) { interleaved, non-interleaved }, "
"rate = " GST_AUDIO_RATE_RANGE ", "
"channels = " GST_AUDIO_CHANNELS_RANGE)
);
#define gst_audio_test_src_parent_class parent_class
G_DEFINE_TYPE (GstAudioTestSrc, gst_audio_test_src, GST_TYPE_BASE_SRC);
#define GST_TYPE_AUDIO_TEST_SRC_WAVE (gst_audiostestsrc_wave_get_type())
static GType
gst_audiostestsrc_wave_get_type (void)
{
static GType audiostestsrc_wave_type = 0;
static const GEnumValue audiostestsrc_waves[] = {
{GST_AUDIO_TEST_SRC_WAVE_SINE, "Sine", "sine"},
{GST_AUDIO_TEST_SRC_WAVE_SQUARE, "Square", "square"},
{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 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_TICKS, "Periodic Ticks", "ticks"},
{GST_AUDIO_TEST_SRC_WAVE_GAUSSIAN_WHITE_NOISE, "White Gaussian noise",
"gaussian-noise"},
{GST_AUDIO_TEST_SRC_WAVE_RED_NOISE, "Red (brownian) noise", "red-noise"},
{GST_AUDIO_TEST_SRC_WAVE_BLUE_NOISE, "Blue noise", "blue-noise"},
{GST_AUDIO_TEST_SRC_WAVE_VIOLET_NOISE, "Violet noise", "violet-noise"},
{0, NULL, NULL},
};
if (G_UNLIKELY (audiostestsrc_wave_type == 0)) {
audiostestsrc_wave_type = g_enum_register_static ("GstAudioTestSrcWave",
audiostestsrc_waves);
}
return audiostestsrc_wave_type;
}
static void gst_audio_test_src_finalize (GObject * object);
static void gst_audio_test_src_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_test_src_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_audio_test_src_setcaps (GstBaseSrc * basesrc,
GstCaps * caps);
static GstCaps *gst_audio_test_src_fixate (GstBaseSrc * bsrc, 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_query (GstBaseSrc * basesrc,
GstQuery * query);
static void gst_audio_test_src_change_wave (GstAudioTestSrc * src);
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_fill (GstBaseSrc * basesrc,
guint64 offset, guint length, GstBuffer * buffer);
static void
gst_audio_test_src_class_init (GstAudioTestSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gobject_class->set_property = gst_audio_test_src_set_property;
gobject_class->get_property = gst_audio_test_src_get_property;
gobject_class->finalize = gst_audio_test_src_finalize;
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, 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. "
"The sample rate needs to be at least 2 times higher.",
0.0, (gdouble) G_MAXINT / 2, 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, 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", 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",
"An offset added to timestamps set on buffers (in ns)", G_MININT64,
G_MAXINT64, DEFAULT_TIMESTAMP_OFFSET,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SINE_PERIODS_PER_TICK,
g_param_spec_uint ("sine-periods-per-tick", "Sine periods per tick",
"Number of sine wave periods in one tick. Only used if wave = ticks.",
1, G_MAXUINT, DEFAULT_SINE_PERIODS_PER_TICK,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_TICK_INTERVAL,
g_param_spec_uint64 ("tick-interval", "Time between ticks",
"Distance between start of current and start of next tick, in nanoseconds.",
1, G_MAXUINT64, DEFAULT_TIME_BETWEEN_TICKS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MARKER_TICK_PERIOD,
g_param_spec_uint ("marker-tick-period", "Marker tick period",
"Make every Nth tick a marker tick (= a tick with different volume). "
"Only used if wave = ticks. 0 = no marker ticks.",
0, G_MAXUINT, DEFAULT_MARKER_TICK_PERIOD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MARKER_TICK_VOLUME,
g_param_spec_double ("marker-tick-volume", "Marker tick volume",
"Volume of marker ticks. Only used if wave = ticks and"
"marker-tick-period is set to a nonzero value.",
0.0, 1.0, DEFAULT_MARKER_TICK_VOLUME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_APPLY_TICK_RAMP,
g_param_spec_boolean ("apply-tick-ramp", "Apply tick ramp",
"Apply ramp to tick samples", DEFAULT_APPLY_TICK_RAMP,
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));
gst_element_class_add_static_pad_template (gstelement_class,
&gst_audio_test_src_src_template);
gst_element_class_set_static_metadata (gstelement_class, "Audio test source",
"Source/Audio",
"Creates audio test signals of given frequency and volume",
"Stefan Kost <ensonic@users.sf.net>");
gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_test_src_setcaps);
gstbasesrc_class->fixate = GST_DEBUG_FUNCPTR (gst_audio_test_src_fixate);
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_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->fill = GST_DEBUG_FUNCPTR (gst_audio_test_src_fill);
}
static void
gst_audio_test_src_init (GstAudioTestSrc * src)
{
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), DEFAULT_IS_LIVE);
src->samples_per_buffer = DEFAULT_SAMPLES_PER_BUFFER;
src->generate_samples_per_buffer = src->samples_per_buffer;
src->timestamp_offset = DEFAULT_TIMESTAMP_OFFSET;
src->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL;
src->sine_periods_per_tick = DEFAULT_SINE_PERIODS_PER_TICK;
src->tick_interval = DEFAULT_TIME_BETWEEN_TICKS;
src->marker_tick_period = DEFAULT_MARKER_TICK_PERIOD;
src->marker_tick_volume = DEFAULT_MARKER_TICK_VOLUME;
src->apply_tick_ramp = DEFAULT_APPLY_TICK_RAMP;
src->gen = NULL;
src->wave = DEFAULT_WAVE;
gst_base_src_set_blocksize (GST_BASE_SRC (src), -1);
}
static void
gst_audio_test_src_finalize (GObject * object)
{
GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (object);
if (src->gen)
g_rand_free (src->gen);
src->gen = NULL;
g_free (src->tmp);
src->tmp = NULL;
src->tmpsize = 0;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstCaps *
gst_audio_test_src_fixate (GstBaseSrc * bsrc, GstCaps * caps)
{
GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (bsrc);
GstStructure *structure;
gint channels, rate;
caps = gst_caps_make_writable (caps);
structure = gst_caps_get_structure (caps, 0);
GST_DEBUG_OBJECT (src, "fixating samplerate to %d", GST_AUDIO_DEF_RATE);
rate = MAX (GST_AUDIO_DEF_RATE, src->freq * 2);
gst_structure_fixate_field_nearest_int (structure, "rate", rate);
gst_structure_fixate_field_string (structure, "format", DEFAULT_FORMAT_STR);
gst_structure_fixate_field_string (structure, "layout", "interleaved");
/* fixate to mono unless downstream requires stereo, for backwards compat */
gst_structure_fixate_field_nearest_int (structure, "channels", 1);
if (gst_structure_get_int (structure, "channels", &channels) && channels > 2) {
if (!gst_structure_has_field_typed (structure, "channel-mask",
GST_TYPE_BITMASK))
gst_structure_set (structure, "channel-mask", GST_TYPE_BITMASK, 0ULL,
NULL);
}
caps = GST_BASE_SRC_CLASS (parent_class)->fixate (bsrc, caps);
return caps;
}
static gboolean
gst_audio_test_src_setcaps (GstBaseSrc * basesrc, GstCaps * caps)
{
GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (basesrc);
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps))
goto invalid_caps;
GST_DEBUG_OBJECT (src, "negotiated to caps %" GST_PTR_FORMAT, caps);
src->info = info;
gst_base_src_set_blocksize (basesrc,
GST_AUDIO_INFO_BPF (&info) * src->samples_per_buffer);
gst_audio_test_src_change_wave (src);
return TRUE;
/* ERROR */
invalid_caps:
{
GST_ERROR_OBJECT (basesrc, "received invalid caps");
return FALSE;
}
}
static gboolean
gst_audio_test_src_query (GstBaseSrc * basesrc, GstQuery * query)
{
GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (basesrc);
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CONVERT:
{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
if (!gst_audio_info_convert (&src->info, src_fmt, src_val, dest_fmt,
&dest_val))
goto error;
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
res = TRUE;
break;
}
case GST_QUERY_SCHEDULING:
{
/* if we can operate in pull mode */
gst_query_set_scheduling (query, GST_SCHEDULING_FLAG_SEEKABLE, 1, -1, 0);
gst_query_add_scheduling_mode (query, GST_PAD_MODE_PUSH);
if (src->can_activate_pull)
gst_query_add_scheduling_mode (query, GST_PAD_MODE_PULL);
res = TRUE;
break;
}
case GST_QUERY_LATENCY:
{
if (src->info.rate > 0) {
GstClockTime latency;
latency =
gst_util_uint64_scale (src->generate_samples_per_buffer, GST_SECOND,
src->info.rate);
gst_query_set_latency (query,
gst_base_src_is_live (GST_BASE_SRC_CAST (src)), latency,
GST_CLOCK_TIME_NONE);
GST_DEBUG_OBJECT (src, "Reporting latency of %" GST_TIME_FORMAT,
GST_TIME_ARGS (latency));
res = TRUE;
}
break;
}
default:
res = GST_BASE_SRC_CLASS (parent_class)->query (basesrc, query);
break;
}
return res;
/* ERROR */
error:
{
GST_DEBUG_OBJECT (src, "query failed");
return FALSE;
}
}
#define DEFINE_SINE(type,scale) \
static void \
gst_audio_test_src_create_sine_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channels, channel_step, sample_step; \
gdouble step, amp; \
g##type *ptr; \
\
channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
channel_step = 1; \
sample_step = channels; \
} else { \
channel_step = src->generate_samples_per_buffer; \
sample_step = 1; \
} \
step = M_PI_M2 * src->freq / GST_AUDIO_INFO_RATE (&src->info); \
amp = src->volume * scale; \
\
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
src->accumulator += step; \
if (src->accumulator >= M_PI_M2) \
src->accumulator -= M_PI_M2; \
\
ptr = samples; \
for (c = 0; c < channels; ++c) { \
*ptr = (g##type) (sin (src->accumulator) * amp); \
ptr += channel_step; \
} \
samples += sample_step; \
} \
}
DEFINE_SINE (int16, 32767.0);
DEFINE_SINE (int32, 2147483647.0);
DEFINE_SINE (float, 1.0);
DEFINE_SINE (double, 1.0);
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,
(ProcessFunc) gst_audio_test_src_create_sine_double
};
#define DEFINE_SQUARE(type,scale) \
static void \
gst_audio_test_src_create_square_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channels, channel_step, sample_step; \
gdouble step, amp; \
g##type *ptr; \
\
channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
channel_step = 1; \
sample_step = channels; \
} else { \
channel_step = src->generate_samples_per_buffer; \
sample_step = 1; \
} \
step = M_PI_M2 * src->freq / GST_AUDIO_INFO_RATE (&src->info); \
amp = src->volume * scale; \
\
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
src->accumulator += step; \
if (src->accumulator >= M_PI_M2) \
src->accumulator -= M_PI_M2; \
\
ptr = samples; \
for (c = 0; c < channels; ++c) { \
*ptr = (g##type) ((src->accumulator < G_PI) ? amp : -amp); \
ptr += channel_step; \
} \
samples += sample_step; \
} \
}
DEFINE_SQUARE (int16, 32767.0);
DEFINE_SQUARE (int32, 2147483647.0);
DEFINE_SQUARE (float, 1.0);
DEFINE_SQUARE (double, 1.0);
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,
(ProcessFunc) gst_audio_test_src_create_square_double
};
#define DEFINE_SAW(type,scale) \
static void \
gst_audio_test_src_create_saw_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channels, channel_step, sample_step; \
gdouble step, amp; \
g##type *ptr; \
\
channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
channel_step = 1; \
sample_step = channels; \
} else { \
channel_step = src->generate_samples_per_buffer; \
sample_step = 1; \
} \
step = M_PI_M2 * src->freq / GST_AUDIO_INFO_RATE (&src->info); \
amp = (src->volume * scale) / G_PI; \
\
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
src->accumulator += step; \
if (src->accumulator >= M_PI_M2) \
src->accumulator -= M_PI_M2; \
\
ptr = samples; \
if (src->accumulator < G_PI) { \
for (c = 0; c < channels; ++c) { \
*ptr = (g##type) (src->accumulator * amp); \
ptr += channel_step; \
} \
} else { \
for (c = 0; c < channels; ++c) { \
*ptr = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
ptr += channel_step; \
} \
} \
samples += sample_step; \
} \
}
DEFINE_SAW (int16, 32767.0);
DEFINE_SAW (int32, 2147483647.0);
DEFINE_SAW (float, 1.0);
DEFINE_SAW (double, 1.0);
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,
(ProcessFunc) gst_audio_test_src_create_saw_double
};
#define DEFINE_TRIANGLE(type,scale) \
static void \
gst_audio_test_src_create_triangle_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channels, channel_step, sample_step; \
gdouble step, amp; \
g##type *ptr; \
\
channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
channel_step = 1; \
sample_step = channels; \
} else { \
channel_step = src->generate_samples_per_buffer; \
sample_step = 1; \
} \
step = M_PI_M2 * src->freq / GST_AUDIO_INFO_RATE (&src->info); \
amp = (src->volume * scale) / G_PI_2; \
\
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
src->accumulator += step; \
if (src->accumulator >= M_PI_M2) \
src->accumulator -= M_PI_M2; \
\
ptr = samples; \
if (src->accumulator < (G_PI_2)) { \
for (c = 0; c < channels; ++c) { \
*ptr = (g##type) (src->accumulator * amp); \
ptr += channel_step; \
} \
} else if (src->accumulator < (G_PI * 1.5)) { \
for (c = 0; c < channels; ++c) { \
*ptr = (g##type) ((src->accumulator - G_PI) * -amp); \
ptr += channel_step; \
} \
} else { \
for (c = 0; c < channels; ++c) { \
*ptr = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
ptr += channel_step; \
} \
} \
samples += sample_step; \
} \
}
DEFINE_TRIANGLE (int16, 32767.0);
DEFINE_TRIANGLE (int32, 2147483647.0);
DEFINE_TRIANGLE (float, 1.0);
DEFINE_TRIANGLE (double, 1.0);
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,
(ProcessFunc) gst_audio_test_src_create_triangle_double
};
#define DEFINE_SILENCE(type) \
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) * src->info.channels); \
}
DEFINE_SILENCE (int16);
DEFINE_SILENCE (int32);
DEFINE_SILENCE (float);
DEFINE_SILENCE (double);
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,
(ProcessFunc) gst_audio_test_src_create_silence_double
};
#define DEFINE_WHITE_NOISE(type,scale) \
static void \
gst_audio_test_src_create_white_noise_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channel_step, sample_step; \
g##type *ptr; \
gdouble amp = (src->volume * scale); \
gint channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
\
if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
channel_step = 1; \
sample_step = channels; \
} else { \
channel_step = src->generate_samples_per_buffer; \
sample_step = 1; \
} \
\
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
ptr = samples; \
for (c = 0; c < channels; ++c) { \
*ptr = (g##type) (amp * g_rand_double_range (src->gen, -1.0, 1.0)); \
ptr += channel_step; \
} \
samples += sample_step; \
} \
}
DEFINE_WHITE_NOISE (int16, 32767.0);
DEFINE_WHITE_NOISE (int32, 2147483647.0);
DEFINE_WHITE_NOISE (float, 1.0);
DEFINE_WHITE_NOISE (double, 1.0);
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,
(ProcessFunc) gst_audio_test_src_create_white_noise_double
};
/* pink noise calculation is based on
* http://www.firstpr.com.au/dsp/pink-noise/phil_burk_19990905_patest_pink.c
* which has been released under public domain
* Many thanks Phil!
*/
static void
gst_audio_test_src_init_pink_noise (GstAudioTestSrc * src)
{
gint i;
gint num_rows = 12; /* arbitrary: 1 .. PINK_MAX_RANDOM_ROWS */
glong pmax;
src->pink.index = 0;
src->pink.index_mask = (1 << num_rows) - 1;
/* calculate maximum possible signed random value.
* Extra 1 for white noise always added. */
pmax = (num_rows + 1) * (1 << (PINK_RANDOM_BITS - 1));
src->pink.scalar = 1.0f / pmax;
/* Initialize rows. */
for (i = 0; i < num_rows; i++)
src->pink.rows[i] = 0;
src->pink.running_sum = 0;
}
/* Generate Pink noise values between -1.0 and +1.0 */
static gdouble
gst_audio_test_src_generate_pink_noise_value (GstAudioTestSrc * src)
{
GstPinkNoise *pink = &src->pink;
glong new_random;
glong sum;
/* Increment and mask index. */
pink->index = (pink->index + 1) & pink->index_mask;
/* If index is zero, don't update any random values. */
if (pink->index != 0) {
/* Determine how many trailing zeros in PinkIndex. */
/* This algorithm will hang if n==0 so test first. */
gint num_zeros = 0;
gint n = pink->index;
while ((n & 1) == 0) {
n = n >> 1;
num_zeros++;
}
/* Replace the indexed ROWS random value.
* Subtract and add back to RunningSum instead of adding all the random
* values together. Only one changes each time.
*/
pink->running_sum -= pink->rows[num_zeros];
new_random = 32768.0 - (65536.0 * (gulong) g_rand_int (src->gen)
/ (G_MAXUINT32 + 1.0));
pink->running_sum += new_random;
pink->rows[num_zeros] = new_random;
}
/* Add extra white noise value. */
new_random = 32768.0 - (65536.0 * (gulong) g_rand_int (src->gen)
/ (G_MAXUINT32 + 1.0));
sum = pink->running_sum + new_random;
/* Scale to range of -1.0 to 0.9999. */
return (pink->scalar * sum);
}
#define DEFINE_PINK(type, scale) \
static void \
gst_audio_test_src_create_pink_noise_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channels, channel_step, sample_step; \
gdouble amp; \
g##type *ptr; \
\
amp = src->volume * scale; \
channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
channel_step = 1; \
sample_step = channels; \
} else { \
channel_step = src->generate_samples_per_buffer; \
sample_step = 1; \
} \
\
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
ptr = samples; \
for (c = 0; c < channels; ++c) { \
*ptr = (g##type) (gst_audio_test_src_generate_pink_noise_value (src) * amp); \
ptr += channel_step; \
} \
samples += sample_step; \
} \
}
DEFINE_PINK (int16, 32767.0);
DEFINE_PINK (int32, 2147483647.0);
DEFINE_PINK (float, 1.0);
DEFINE_PINK (double, 1.0);
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,
(ProcessFunc) gst_audio_test_src_create_pink_noise_double
};
static void
gst_audio_test_src_init_sine_table (GstAudioTestSrc * src, gboolean use_volume)
{
gint i;
gdouble ang = 0.0;
gdouble step = M_PI_M2 / 1024.0;
gdouble amp = use_volume ? src->volume : 1.0;
for (i = 0; i < 1024; i++) {
src->wave_table[i] = sin (ang) * amp;
ang += step;
}
}
#define DEFINE_SINE_TABLE(type,scale) \
static void \
gst_audio_test_src_create_sine_table_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channels, channel_step, sample_step; \
gdouble step, scl; \
g##type *ptr; \
\
channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
channel_step = 1; \
sample_step = channels; \
} else { \
channel_step = src->generate_samples_per_buffer; \
sample_step = 1; \
} \
step = M_PI_M2 * src->freq / GST_AUDIO_INFO_RATE (&src->info); \
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; \
\
ptr = samples; \
for (c = 0; c < channels; ++c) { \
*ptr = (g##type) scale * src->wave_table[(gint) (src->accumulator * scl)]; \
ptr += channel_step; \
} \
samples += sample_step; \
} \
}
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
};
static inline gdouble
calc_scaled_tick_volume (GstAudioTestSrc * src, gdouble scale)
{
gdouble vol;
vol = ((src->marker_tick_period > 0)
&& ((src->tick_counter % src->marker_tick_period) == 0))
? src->marker_tick_volume : src->volume;
return vol * scale;
}
#define DEFINE_TICKS(type,scale) \
static void \
gst_audio_test_src_create_tick_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channels, samplerate, samplemod, channel_step, sample_step; \
gint num_nonzero_samples, num_ramp_samples, end_ramp_offset; \
gdouble step, scl; \
gdouble volscale; \
g##type *ptr; \
\
volscale = calc_scaled_tick_volume (src, scale); \
channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
channel_step = 1; \
sample_step = channels; \
} else { \
channel_step = src->generate_samples_per_buffer; \
sample_step = 1; \
} \
samplerate = GST_AUDIO_INFO_RATE (&src->info); \
step = M_PI_M2 * src->freq / samplerate; \
num_nonzero_samples = samplerate * src->sine_periods_per_tick / src->freq; \
scl = 1024.0 / M_PI_M2; \
num_ramp_samples = src->apply_tick_ramp ? (samplerate / src->freq) : 0; \
end_ramp_offset = num_nonzero_samples - num_ramp_samples; \
\
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
samplemod = (src->next_sample + i)%src->samples_between_ticks; \
\
ptr = samples; \
if (samplemod == 0) { \
src->accumulator = 0; \
src->tick_counter++; \
volscale = calc_scaled_tick_volume (src, scale); \
} else if (samplemod < num_nonzero_samples) { \
gdouble ramp; \
if (num_ramp_samples > 0) { \
ramp = \
(samplemod < num_ramp_samples) ? (((gdouble)samplemod) / num_ramp_samples) : \
(samplemod >= end_ramp_offset) ? (((gdouble)(num_nonzero_samples - samplemod)) / num_ramp_samples) \
: 1.0; \
if (ramp > 1.0) \
ramp = 1.0; \
ramp *= ramp * ramp; \
} else \
ramp = 1.0; \
\
for (c = 0; c < channels; ++c) { \
*ptr = \
(g##type) volscale * ramp * src->wave_table[(gint) (src->accumulator * scl)]; \
ptr += channel_step; \
} \
} else { \
for (c = 0; c < channels; ++c) { \
*ptr = 0; \
ptr += channel_step; \
} \
} \
\
src->accumulator += step; \
if (src->accumulator >= M_PI_M2) \
src->accumulator -= M_PI_M2; \
\
samples += sample_step; \
} \
}
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 complex random variable with
* uniformly-distributed argument and \chi^{2}-distributed modulus.
*/
#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, channel_step, sample_step; \
g##type *ptr; \
gdouble amp = (src->volume * scale); \
gint channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
\
if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
channel_step = 1; \
sample_step = channels; \
} else { \
channel_step = src->generate_samples_per_buffer; \
sample_step = 1; \
} \
\
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
ptr = samples; \
for (c = 0; c < channels; ++c) { \
gdouble mag = sqrt (-2 * log (1.0 - g_rand_double (src->gen))); \
gdouble phs = g_rand_double_range (src->gen, 0.0, M_PI_M2); \
\
*ptr = (g##type) (amp * mag * cos (phs)); \
ptr += channel_step; \
if (++c >= channels) \
break; \
*ptr = (g##type) (amp * mag * sin (phs)); \
ptr += channel_step; \
} \
samples += sample_step; \
} \
}
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
};
/* Brownian (Red) Noise: noise where the power density decreases by 6 dB per
* octave with increasing frequency
*
* taken from http://vellocet.com/dsp/noise/VRand.html
* by Andrew Simper of Vellocet (andy@vellocet.com)
*/
#define DEFINE_RED_NOISE(type,scale) \
static void \
gst_audio_test_src_create_red_noise_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channel_step, sample_step; \
g##type *ptr; \
gdouble amp = (src->volume * scale); \
gdouble state = src->red.state; \
gint channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
\
if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
channel_step = 1; \
sample_step = channels; \
} else { \
channel_step = src->generate_samples_per_buffer; \
sample_step = 1; \
} \
\
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
ptr = samples; \
for (c = 0; c < channels; ++c) { \
while (TRUE) { \
gdouble r = g_rand_double_range (src->gen, -1.0, 1.0); \
state += r; \
if (state < -8.0f || state > 8.0f) state -= r; \
else break; \
} \
*ptr = (g##type) (amp * state * 0.0625f); /* /16.0 */ \
ptr += channel_step; \
} \
samples += sample_step; \
} \
src->red.state = state; \
}
DEFINE_RED_NOISE (int16, 32767.0);
DEFINE_RED_NOISE (int32, 2147483647.0);
DEFINE_RED_NOISE (float, 1.0);
DEFINE_RED_NOISE (double, 1.0);
static const ProcessFunc red_noise_funcs[] = {
(ProcessFunc) gst_audio_test_src_create_red_noise_int16,
(ProcessFunc) gst_audio_test_src_create_red_noise_int32,
(ProcessFunc) gst_audio_test_src_create_red_noise_float,
(ProcessFunc) gst_audio_test_src_create_red_noise_double
};
/* Blue Noise: apply spectral inversion to pink noise */
#define DEFINE_BLUE_NOISE(type) \
static void \
gst_audio_test_src_create_blue_noise_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channel_step, sample_step; \
static gdouble flip=1.0; \
gint channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
g##type *ptr; \
\
if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
channel_step = 1; \
sample_step = channels; \
} else { \
channel_step = src->generate_samples_per_buffer; \
sample_step = 1; \
} \
\
gst_audio_test_src_create_pink_noise_##type (src, samples); \
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
ptr = samples; \
for (c = 0; c < channels; ++c) { \
*ptr *= flip; \
ptr += channel_step; \
} \
flip *= -1.0; \
samples += sample_step; \
} \
}
DEFINE_BLUE_NOISE (int16);
DEFINE_BLUE_NOISE (int32);
DEFINE_BLUE_NOISE (float);
DEFINE_BLUE_NOISE (double);
static const ProcessFunc blue_noise_funcs[] = {
(ProcessFunc) gst_audio_test_src_create_blue_noise_int16,
(ProcessFunc) gst_audio_test_src_create_blue_noise_int32,
(ProcessFunc) gst_audio_test_src_create_blue_noise_float,
(ProcessFunc) gst_audio_test_src_create_blue_noise_double
};
/* Violet Noise: apply spectral inversion to red noise */
#define DEFINE_VIOLET_NOISE(type) \
static void \
gst_audio_test_src_create_violet_noise_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channel_step, sample_step; \
static gdouble flip=1.0; \
gint channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
g##type *ptr; \
\
if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
channel_step = 1; \
sample_step = channels; \
} else { \
channel_step = src->generate_samples_per_buffer; \
sample_step = 1; \
} \
\
gst_audio_test_src_create_red_noise_##type (src, samples); \
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
ptr = samples; \
for (c = 0; c < channels; ++c) { \
*ptr *= flip; \
ptr += channel_step; \
} \
flip *= -1.0; \
samples += sample_step; \
} \
}
DEFINE_VIOLET_NOISE (int16);
DEFINE_VIOLET_NOISE (int32);
DEFINE_VIOLET_NOISE (float);
DEFINE_VIOLET_NOISE (double);
static const ProcessFunc violet_noise_funcs[] = {
(ProcessFunc) gst_audio_test_src_create_violet_noise_int16,
(ProcessFunc) gst_audio_test_src_create_violet_noise_int32,
(ProcessFunc) gst_audio_test_src_create_violet_noise_float,
(ProcessFunc) gst_audio_test_src_create_violet_noise_double
};
/*
* gst_audio_test_src_change_wave:
* Assign function pointer of wave generator.
*/
static void
gst_audio_test_src_change_wave (GstAudioTestSrc * src)
{
gint idx;
src->pack_func = NULL;
src->process = NULL;
/* not negotiated yet? */
if (src->info.finfo == NULL)
return;
switch (GST_AUDIO_FORMAT_INFO_FORMAT (src->info.finfo)) {
case GST_AUDIO_FORMAT_S16:
idx = 0;
break;
case GST_AUDIO_FORMAT_S32:
idx = 1;
break;
case GST_AUDIO_FORMAT_F32:
idx = 2;
break;
case GST_AUDIO_FORMAT_F64:
idx = 3;
break;
default:
/* special format */
switch (src->info.finfo->unpack_format) {
case GST_AUDIO_FORMAT_S32:
idx = 1;
src->pack_func = src->info.finfo->pack_func;
src->pack_size = sizeof (gint32);
break;
case GST_AUDIO_FORMAT_F64:
idx = 3;
src->pack_func = src->info.finfo->pack_func;
src->pack_size = sizeof (gdouble);
break;
default:
g_assert_not_reached ();
return;
}
}
switch (src->wave) {
case GST_AUDIO_TEST_SRC_WAVE_SINE:
src->process = sine_funcs[idx];
break;
case GST_AUDIO_TEST_SRC_WAVE_SQUARE:
src->process = square_funcs[idx];
break;
case GST_AUDIO_TEST_SRC_WAVE_SAW:
src->process = saw_funcs[idx];
break;
case GST_AUDIO_TEST_SRC_WAVE_TRIANGLE:
src->process = triangle_funcs[idx];
break;
case GST_AUDIO_TEST_SRC_WAVE_SILENCE:
src->process = silence_funcs[idx];
break;
case GST_AUDIO_TEST_SRC_WAVE_WHITE_NOISE:
if (!(src->gen))
src->gen = g_rand_new ();
src->process = white_noise_funcs[idx];
break;
case GST_AUDIO_TEST_SRC_WAVE_PINK_NOISE:
if (!(src->gen))
src->gen = g_rand_new ();
gst_audio_test_src_init_pink_noise (src);
src->process = pink_noise_funcs[idx];
break;
case GST_AUDIO_TEST_SRC_WAVE_SINE_TAB:
gst_audio_test_src_init_sine_table (src, TRUE);
src->process = sine_table_funcs[idx];
break;
case GST_AUDIO_TEST_SRC_WAVE_TICKS:
gst_audio_test_src_init_sine_table (src, FALSE);
src->process = tick_funcs[idx];
src->samples_between_ticks =
gst_util_uint64_scale_int (src->tick_interval,
GST_AUDIO_INFO_RATE (&(src->info)), GST_SECOND);
break;
case GST_AUDIO_TEST_SRC_WAVE_GAUSSIAN_WHITE_NOISE:
if (!(src->gen))
src->gen = g_rand_new ();
src->process = gaussian_white_noise_funcs[idx];
break;
case GST_AUDIO_TEST_SRC_WAVE_RED_NOISE:
if (!(src->gen))
src->gen = g_rand_new ();
src->red.state = 0.0;
src->process = red_noise_funcs[idx];
break;
case GST_AUDIO_TEST_SRC_WAVE_BLUE_NOISE:
if (!(src->gen))
src->gen = g_rand_new ();
gst_audio_test_src_init_pink_noise (src);
src->process = blue_noise_funcs[idx];
break;
case GST_AUDIO_TEST_SRC_WAVE_VIOLET_NOISE:
if (!(src->gen))
src->gen = g_rand_new ();
src->red.state = 0.0;
src->process = violet_noise_funcs[idx];
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, TRUE);
break;
default:
break;
}
}
static void
gst_audio_test_src_get_times (GstBaseSrc * basesrc, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
/* for live sources, sync on the timestamp of the buffer */
if (gst_base_src_is_live (basesrc)) {
GstClockTime timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
/* get duration to calculate end time */
GstClockTime duration = GST_BUFFER_DURATION (buffer);
if (GST_CLOCK_TIME_IS_VALID (duration)) {
*end = timestamp + duration;
}
*start = timestamp;
}
} else {
*start = -1;
*end = -1;
}
}
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;
src->tick_counter = 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);
GstClockTime time;
gint samplerate, bpf;
gint64 next_sample;
GST_DEBUG_OBJECT (src, "seeking %" GST_SEGMENT_FORMAT, segment);
time = segment->position;
src->reverse = (segment->rate < 0.0);
samplerate = GST_AUDIO_INFO_RATE (&src->info);
bpf = GST_AUDIO_INFO_BPF (&src->info);
/* now move to the time indicated, don't seek to the sample *after* the time */
next_sample = gst_util_uint64_scale_int (time, samplerate, GST_SECOND);
src->next_byte = next_sample * bpf;
if (samplerate == 0)
src->next_time = 0;
else
src->next_time =
gst_util_uint64_scale_round (next_sample, GST_SECOND, samplerate);
GST_DEBUG_OBJECT (src, "seeking next_sample=%" G_GINT64_FORMAT
" next_time=%" GST_TIME_FORMAT, next_sample,
GST_TIME_ARGS (src->next_time));
g_assert (src->next_time <= time);
src->next_sample = next_sample;
if (segment->rate > 0 && GST_CLOCK_TIME_IS_VALID (segment->stop)) {
time = segment->stop;
src->sample_stop =
gst_util_uint64_scale_round (time, samplerate, GST_SECOND);
src->check_seek_stop = TRUE;
} else if (segment->rate < 0) {
time = segment->start;
src->sample_stop =
gst_util_uint64_scale_round (time, samplerate, GST_SECOND);
src->check_seek_stop = TRUE;
} else {
src->check_seek_stop = FALSE;
}
src->eos_reached = FALSE;
return TRUE;
}
static gboolean
gst_audio_test_src_is_seekable (GstBaseSrc * basesrc)
{
/* we're seekable... */
return TRUE;
}
static GstFlowReturn
gst_audio_test_src_fill (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer * buffer)
{
GstAudioTestSrc *src;
GstClockTime next_time;
gint64 next_sample, next_byte;
gint bytes, samples;
GstElementClass *eclass;
GstMapInfo map;
gint samplerate, bpf;
src = GST_AUDIO_TEST_SRC (basesrc);
/* example for tagging generated data */
if (!src->tags_pushed) {
GstTagList *taglist;
taglist = gst_tag_list_new (GST_TAG_DESCRIPTION, "audiotest wave", NULL);
eclass = GST_ELEMENT_CLASS (parent_class);
if (eclass->send_event)
eclass->send_event (GST_ELEMENT_CAST (basesrc),
gst_event_new_tag (taglist));
else
gst_tag_list_unref (taglist);
src->tags_pushed = TRUE;
}
if (src->eos_reached) {
GST_INFO_OBJECT (src, "eos");
return GST_FLOW_EOS;
}
samplerate = GST_AUDIO_INFO_RATE (&src->info);
bpf = GST_AUDIO_INFO_BPF (&src->info);
/* 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 / bpf;
/* 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 / bpf;
src->next_time =
gst_util_uint64_scale_int (src->next_sample, GST_SECOND, samplerate);
src->next_byte = offset;
}
/* check for eos */
if (src->check_seek_stop && !src->reverse &&
(src->sample_stop > src->next_sample) &&
(src->sample_stop < src->next_sample + samples)
) {
/* calculate only partial buffer */
src->generate_samples_per_buffer = src->sample_stop - src->next_sample;
next_sample = src->sample_stop;
src->eos_reached = TRUE;
} else if (src->check_seek_stop && src->reverse &&
(src->sample_stop > src->next_sample)
) {
/* calculate only partial buffer */
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 = samples;
next_sample = src->next_sample + (src->reverse ? (-samples) : samples);
}
bytes = src->generate_samples_per_buffer * bpf;
next_byte = src->next_byte + (src->reverse ? (-bytes) : bytes);
next_time = gst_util_uint64_scale_int (next_sample, GST_SECOND, samplerate);
GST_LOG_OBJECT (src, "samplerate %d", samplerate);
GST_LOG_OBJECT (src, "next_sample %" G_GINT64_FORMAT ", ts %" GST_TIME_FORMAT,
next_sample, GST_TIME_ARGS (next_time));
gst_buffer_set_size (buffer, bytes);
GST_BUFFER_OFFSET (buffer) = src->next_sample;
GST_BUFFER_OFFSET_END (buffer) = next_sample;
if (!src->reverse) {
GST_BUFFER_TIMESTAMP (buffer) = src->timestamp_offset + src->next_time;
GST_BUFFER_DURATION (buffer) = next_time - src->next_time;
} else {
GST_BUFFER_TIMESTAMP (buffer) = src->timestamp_offset + next_time;
GST_BUFFER_DURATION (buffer) = src->next_time - next_time;
}
gst_object_sync_values (GST_OBJECT (src), GST_BUFFER_TIMESTAMP (buffer));
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,
src->generate_samples_per_buffer,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
if (src->pack_func) {
gsize tmpsize;
tmpsize =
src->generate_samples_per_buffer * GST_AUDIO_INFO_CHANNELS (&src->info)
* src->pack_size;
if (tmpsize > src->tmpsize) {
src->tmp = g_realloc (src->tmp, tmpsize);
src->tmpsize = tmpsize;
}
src->process (src, src->tmp);
src->pack_func (src->info.finfo, 0, src->tmp, map.data,
src->generate_samples_per_buffer *
GST_AUDIO_INFO_CHANNELS (&src->info));
} else {
src->process (src, map.data);
}
gst_buffer_unmap (buffer, &map);
if (G_UNLIKELY ((src->wave == GST_AUDIO_TEST_SRC_WAVE_SILENCE)
|| (src->volume == 0.0))) {
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_GAP);
}
if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
gst_buffer_add_audio_meta (buffer, &src->info,
src->generate_samples_per_buffer, NULL);
}
return GST_FLOW_OK;
}
static void
gst_audio_test_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (object);
switch (prop_id) {
case PROP_SAMPLES_PER_BUFFER:
src->samples_per_buffer = g_value_get_int (value);
gst_base_src_set_blocksize (GST_BASE_SRC_CAST (src),
GST_AUDIO_INFO_BPF (&src->info) * src->samples_per_buffer);
break;
case PROP_WAVE:
src->wave = g_value_get_enum (value);
gst_audio_test_src_change_wave (src);
break;
case PROP_FREQ:
src->freq = g_value_get_double (value);
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));
break;
case PROP_TIMESTAMP_OFFSET:
src->timestamp_offset = g_value_get_int64 (value);
break;
case PROP_SINE_PERIODS_PER_TICK:
src->sine_periods_per_tick = g_value_get_uint (value);
break;
case PROP_TICK_INTERVAL:
src->tick_interval = g_value_get_uint64 (value);
break;
case PROP_MARKER_TICK_PERIOD:
src->marker_tick_period = g_value_get_uint (value);
break;
case PROP_MARKER_TICK_VOLUME:
src->marker_tick_volume = g_value_get_double (value);
break;
case PROP_APPLY_TICK_RAMP:
src->apply_tick_ramp = g_value_get_boolean (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;
}
}
static void
gst_audio_test_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (object);
switch (prop_id) {
case PROP_SAMPLES_PER_BUFFER:
g_value_set_int (value, src->samples_per_buffer);
break;
case PROP_WAVE:
g_value_set_enum (value, src->wave);
break;
case PROP_FREQ:
g_value_set_double (value, src->freq);
break;
case PROP_VOLUME:
g_value_set_double (value, src->volume);
break;
case PROP_IS_LIVE:
g_value_set_boolean (value, gst_base_src_is_live (GST_BASE_SRC (src)));
break;
case PROP_TIMESTAMP_OFFSET:
g_value_set_int64 (value, src->timestamp_offset);
break;
case PROP_SINE_PERIODS_PER_TICK:
g_value_set_uint (value, src->sine_periods_per_tick);
break;
case PROP_TICK_INTERVAL:
g_value_set_uint64 (value, src->tick_interval);
break;
case PROP_MARKER_TICK_PERIOD:
g_value_set_uint (value, src->marker_tick_period);
break;
case PROP_MARKER_TICK_VOLUME:
g_value_set_double (value, src->marker_tick_volume);
break;
case PROP_APPLY_TICK_RAMP:
g_value_set_boolean (value, src->apply_tick_ramp);
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;
}
}
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (audio_test_src_debug, "audiotestsrc", 0,
"Audio Test Source");
return gst_element_register (plugin, "audiotestsrc",
GST_RANK_NONE, GST_TYPE_AUDIO_TEST_SRC);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
audiotestsrc,
"Creates audio test signals of given frequency and volume",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
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