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Diffstat (limited to 'gst/audiotestsrc/gstaudiotestsrc.c')
| -rw-r--r-- | gst/audiotestsrc/gstaudiotestsrc.c | 1192 |
1 files changed, 0 insertions, 1192 deletions
diff --git a/gst/audiotestsrc/gstaudiotestsrc.c b/gst/audiotestsrc/gstaudiotestsrc.c deleted file mode 100644 index 2abd41a2..00000000 --- a/gst/audiotestsrc/gstaudiotestsrc.c +++ /dev/null @@ -1,1192 +0,0 @@ -/* 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ -/** - * 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> - * <title>Example launch line</title> - * |[ - * gst-launch audiotestsrc ! audioconvert ! alsasink - * ]| 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 - * ]| 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. - * </refsect2> - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include <math.h> -#include <stdlib.h> -#include <string.h> -#include <gst/controller/gstcontroller.h> - -#include "gstaudiotestsrc.h" - - -#ifndef M_PI -#define M_PI 3.14159265358979323846 -#endif - -#ifndef M_PI_2 -#define M_PI_2 1.57079632679489661923 -#endif - -#define M_PI_M2 ( M_PI + M_PI ) - -GST_DEBUG_CATEGORY_STATIC (audio_test_src_debug); -#define GST_CAT_DEFAULT audio_test_src_debug - -static const GstElementDetails gst_audio_test_src_details = -GST_ELEMENT_DETAILS ("Audio test source", - "Source/Audio", - "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 -{ - PROP_0, - PROP_SAMPLES_PER_BUFFER, - PROP_WAVE, - PROP_FREQ, - PROP_VOLUME, - PROP_IS_LIVE, - PROP_TIMESTAMP_OFFSET, - PROP_CAN_ACTIVATE_PUSH, - PROP_CAN_ACTIVATE_PULL, - PROP_LAST -}; - - -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-int, " - "endianness = (int) BYTE_ORDER, " - "signed = (boolean) true, " - "width = (int) 16, " - "depth = (int) 16, " - "rate = (int) [ 1, MAX ], " - "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, 2 ]; " - "audio/x-raw-float, " - "endianness = (int) BYTE_ORDER, " - "width = (int) { 32, 64 }, " - "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]") - ); - - -GST_BOILERPLATE (GstAudioTestSrc, gst_audio_test_src, GstBaseSrc, - 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"}, - {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_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 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, - 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_create (GstBaseSrc * basesrc, - guint64 offset, guint length, GstBuffer ** buffer); - - -static void -gst_audio_test_src_base_init (gpointer g_class) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&gst_audio_test_src_src_template)); - gst_element_class_set_details (element_class, &gst_audio_test_src_details); -} - -static void -gst_audio_test_src_class_init (GstAudioTestSrcClass * klass) -{ - GObjectClass *gobject_class; - GstBaseSrcClass *gstbasesrc_class; - - gobject_class = (GObjectClass *) 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; - - 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", - 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, 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_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); -} - -static void -gst_audio_test_src_init (GstAudioTestSrc * src, GstAudioTestSrcClass * g_class) -{ - GstPad *pad = GST_BASE_SRC_PAD (src); - - gst_pad_set_fixatecaps_function (pad, gst_audio_test_src_src_fixate); - - src->samplerate = 44100; - src->format = GST_AUDIO_TEST_SRC_FORMAT_NONE; - - 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->wave = DEFAULT_WAVE; - gst_base_src_set_blocksize (GST_BASE_SRC (src), -1); -} - -static void -gst_audio_test_src_src_fixate (GstPad * pad, GstCaps * caps) -{ - GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (GST_PAD_PARENT (pad)); - const gchar *name; - GstStructure *structure; - - 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); - if (strcmp (name, "audio/x-raw-int") == 0) - 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 -gst_audio_test_src_setcaps (GstBaseSrc * basesrc, GstCaps * caps) -{ - GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (basesrc); - const GstStructure *structure; - const gchar *name; - gint width; - gboolean ret; - - 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); - src->format = (width == 32) ? GST_AUDIO_TEST_SRC_FORMAT_S32 : - GST_AUDIO_TEST_SRC_FORMAT_S16; - } else { - ret &= gst_structure_get_int (structure, "width", &width); - src->format = (width == 32) ? GST_AUDIO_TEST_SRC_FORMAT_F32 : - 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; -} - -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 (src_fmt == dest_fmt) { - dest_val = src_val; - goto done; - } - - switch (src_fmt) { - case GST_FORMAT_DEFAULT: - switch (dest_fmt) { - case GST_FORMAT_TIME: - /* samples to time */ - dest_val = - gst_util_uint64_scale_int (src_val, GST_SECOND, - src->samplerate); - break; - default: - goto error; - } - break; - case GST_FORMAT_TIME: - switch (dest_fmt) { - case GST_FORMAT_DEFAULT: - /* time to samples */ - dest_val = - gst_util_uint64_scale_int (src_val, src->samplerate, - GST_SECOND); - break; - default: - goto error; - } - break; - default: - goto error; - } - done: - gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val); - 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; \ - gdouble step, amp; \ - \ - step = M_PI_M2 * src->freq / src->samplerate; \ - amp = src->volume * scale; \ - \ - 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) (sin (src->accumulator) * amp); \ - } \ - } \ -} - -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; \ - gdouble step, amp; \ - \ - step = M_PI_M2 * src->freq / src->samplerate; \ - amp = src->volume * scale; \ - \ - 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) ((src->accumulator < M_PI) ? amp : -amp); \ - } \ - } \ -} - -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; \ - gdouble step, amp; \ - \ - step = M_PI_M2 * src->freq / src->samplerate; \ - amp = (src->volume * scale) / M_PI; \ - \ - 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) { \ - for (c = 0; c < src->channels; ++c) \ - samples[i++] = (g##type) (src->accumulator * amp); \ - } else { \ - for (c = 0; c < src->channels; ++c) \ - samples[i++] = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \ - } \ - } \ -} - -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; \ - gdouble step, amp; \ - \ - step = M_PI_M2 * src->freq / src->samplerate; \ - amp = (src->volume * scale) / M_PI_2; \ - \ - 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)) { \ - for (c = 0; c < src->channels; ++c) \ - samples[i++] = (g##type) (src->accumulator * amp); \ - } else if (src->accumulator < (M_PI * 1.5)) { \ - for (c = 0; c < src->channels; ++c) \ - samples[i++] = (g##type) ((src->accumulator - M_PI) * -amp); \ - } else { \ - for (c = 0; c < src->channels; ++c) \ - samples[i++] = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \ - } \ - } \ -} - -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->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; \ - gdouble amp = (src->volume * scale); \ - \ - 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)); \ - } \ -} - -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 (GstPinkNoise * 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) rand () / (RAND_MAX + 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) rand () / (RAND_MAX + 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; \ - gdouble amp; \ - \ - amp = src->volume * scale; \ - \ - 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); \ - } \ - } \ -} - -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) -{ - gint i; - gdouble ang = 0.0; - gdouble step = M_PI_M2 / 1024.0; - gdouble amp = src->volume; - - 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; \ - 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; \ - 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; \ - \ - 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_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. - */ - -#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 -}; - -/* - * gst_audio_test_src_change_wave: - * Assign function pointer of wave genrator. - */ -static void -gst_audio_test_src_change_wave (GstAudioTestSrc * src) -{ - if (src->format == -1) { - src->process = NULL; - return; - } - - switch (src->wave) { - case GST_AUDIO_TEST_SRC_WAVE_SINE: - src->process = sine_funcs[src->format]; - break; - case GST_AUDIO_TEST_SRC_WAVE_SQUARE: - src->process = square_funcs[src->format]; - break; - case GST_AUDIO_TEST_SRC_WAVE_SAW: - src->process = saw_funcs[src->format]; - break; - case GST_AUDIO_TEST_SRC_WAVE_TRIANGLE: - src->process = triangle_funcs[src->format]; - break; - case GST_AUDIO_TEST_SRC_WAVE_SILENCE: - src->process = silence_funcs[src->format]; - break; - case GST_AUDIO_TEST_SRC_WAVE_WHITE_NOISE: - src->process = white_noise_funcs[src->format]; - break; - case GST_AUDIO_TEST_SRC_WAVE_PINK_NOISE: - gst_audio_test_src_init_pink_noise (src); - src->process = pink_noise_funcs[src->format]; - break; - case GST_AUDIO_TEST_SRC_WAVE_SINE_TAB: - 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; - } -} - -/* - * 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) -{ - /* 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; - - 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; - - segment->time = segment->start; - time = segment->last_stop; - - /* now move to the time indicated */ - src->next_sample = - gst_util_uint64_scale_int (time, src->samplerate, GST_SECOND); - 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->next_time <= time); - - if (GST_CLOCK_TIME_IS_VALID (segment->stop)) { - time = segment->stop; - src->sample_stop = gst_util_uint64_scale_int (time, src->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 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) -{ - GstFlowReturn res; - GstAudioTestSrc *src; - GstBuffer *buf; - GstClockTime next_time; - gint64 next_sample, next_byte; - guint bytes, samples; - GstElementClass *eclass; - - src = GST_AUDIO_TEST_SRC (basesrc); - - /* example for tagging generated data */ - if (!src->tags_pushed) { - GstTagList *taglist; - - taglist = gst_tag_list_new (); - - gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, - 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)); - 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->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 { - /* calculate full buffer */ - src->generate_samples_per_buffer = samples; - next_sample = src->next_sample + samples; - } - - 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; - } - - 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_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; - - 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, - src->generate_samples_per_buffer, - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf))); - - src->process (src, GST_BUFFER_DATA (buf)); - - if (G_UNLIKELY ((src->wave == GST_AUDIO_TEST_SRC_WAVE_SILENCE) - || (src->volume == 0.0))) { - GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_GAP); - } - - *buffer = buf; - - 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); - 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_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_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) -{ - /* initialize gst controller library */ - gst_controller_init (NULL, NULL); - - 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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