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-rw-r--r--src/libsample/Makefile.am6
-rw-r--r--src/libsample/sample.c64
-rw-r--r--src/libsample/sample.h8
3 files changed, 78 insertions, 0 deletions
diff --git a/src/libsample/Makefile.am b/src/libsample/Makefile.am
new file mode 100644
index 0000000..5ae865c
--- /dev/null
+++ b/src/libsample/Makefile.am
@@ -0,0 +1,6 @@
+AM_CPPFLAGS = -Wall -Wextra -g $(all_includes)
+
+noinst_LIBRARIES = libsample.a
+
+libsample_a_SOURCES = \
+ sample.c
diff --git a/src/libsample/sample.c b/src/libsample/sample.c
new file mode 100644
index 0000000..72ba941
--- /dev/null
+++ b/src/libsample/sample.c
@@ -0,0 +1,64 @@
+/* Sample definition
+ *
+ * (C) 2017 by Andreas Eversberg <jolly@eversberg.eu>
+ * All Rights Reserved
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <stdint.h>
+#include "sample.h"
+
+/*
+ * A regular voice conversation takes place at this factor below the full range
+ * of 16 bits signed value:
+ */
+static double int_16_speech_level = SPEECH_LEVEL * 0.7079; /* 16 dBm below dBm0, which is about 3dBm below full 16 bit range */
+
+/* A sample_t is a value that has virtually infinite precision but will also
+ * support high numbers. 'double' or 'float' types are sufficient.
+ *
+ * When using sample_t inside signal processing of each base station, the
+ * level of +- 1 is relative to the normal speach evenlope.
+ *
+ * When converting sample_t to int16_t, the level of +- 1 is reduced by factor.
+ * This way the speech may be louder before clipping happens.
+ *
+ * When using sample_t to modulate (SDR or sound card), the level is changed,
+ * so it represents the frequency deviation in Hz. The deviation of speech
+ * envelope is network dependant.
+ */
+
+void samples_to_int16(int16_t *spl, sample_t *samples, int length)
+{
+ int32_t value;
+
+ while (length--) {
+ value = *samples++ * int_16_speech_level * 32768.0;
+ if (value > 32767.0)
+ *spl++ = 32767;
+ else if (value < -32767.0)
+ *spl++ = -32767;
+ else
+ *spl++ = (uint16_t)value;
+ }
+}
+
+void int16_to_samples(sample_t *samples, int16_t *spl, int length)
+{
+ while (length--) {
+ *samples++ = (double)(*spl++) / 32767.0 / int_16_speech_level;
+ }
+}
+
diff --git a/src/libsample/sample.h b/src/libsample/sample.h
new file mode 100644
index 0000000..01a17df
--- /dev/null
+++ b/src/libsample/sample.h
@@ -0,0 +1,8 @@
+
+typedef double sample_t;
+
+#define SPEECH_LEVEL 0.1585
+
+void samples_to_int16(int16_t *spl, sample_t *samples, int length);
+void int16_to_samples(sample_t *samples, int16_t *spl, int length);
+