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-rw-r--r--tderadio3/src/soundformat.cpp270
1 files changed, 270 insertions, 0 deletions
diff --git a/tderadio3/src/soundformat.cpp b/tderadio3/src/soundformat.cpp
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+++ b/tderadio3/src/soundformat.cpp
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+/***************************************************************************
+ soundformat.cpp - description
+ -------------------
+ begin : Sun Aug 1 2004
+ copyright : (C) 2004 by Martin Witte
+ email : witte@kawo1.rwth-aachen.de
+ ***************************************************************************/
+
+/***************************************************************************
+ * *
+ * 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 2 of the License, or *
+ * (at your option) any later version. *
+ * *
+ ***************************************************************************/
+
+#include "include/soundformat.h"
+
+int SoundFormat::sampleSize() const
+{
+ if (m_SampleBits <= 8) return 1;
+ if (m_SampleBits <= 16) return 2;
+ if (m_SampleBits <= 32) return 4;
+
+ // unknown
+ return -1;
+}
+
+
+int SoundFormat::frameSize() const
+{
+ return sampleSize() * m_Channels;
+}
+
+
+int SoundFormat::minValue() const
+{
+ if (!m_IsSigned) return 0;
+ return -(1 << (m_SampleBits - 1));
+}
+
+
+int SoundFormat::maxValue() const
+{
+ return (1 << (m_SampleBits - m_IsSigned)) - 1;
+}
+
+
+void SoundFormat::restoreConfig(const TQString &prefix, TDEConfig *c)
+{
+ m_SampleBits = c->readNumEntry (prefix + "bits", 16);
+ m_IsSigned = c->readBoolEntry(prefix + "sign", true);
+ m_Channels = c->readNumEntry (prefix + "channels", 2);
+ m_SampleRate = c->readNumEntry (prefix + "samplerate", 44100);
+ bool littleEndian = c->readBoolEntry(prefix + "littleEndian", true);
+ m_Endianess = littleEndian ? LITTLE_ENDIAN : BIG_ENDIAN;
+ m_Encoding = c->readEntry(prefix + "encoding", "raw");
+}
+
+
+void SoundFormat::saveConfig(const TQString &prefix, TDEConfig *c) const
+{
+ c->writeEntry(prefix + "bits", m_SampleBits);
+ c->writeEntry(prefix + "sign", m_IsSigned);
+ c->writeEntry(prefix + "channels", m_Channels);
+ c->writeEntry(prefix + "samplerate", m_SampleRate);
+ c->writeEntry(prefix + "littleEndian", m_Endianess == LITTLE_ENDIAN);
+ c->writeEntry(prefix + "encoding", m_Encoding);
+}
+
+
+int SoundFormat::convertSampleToInt(const char *sample, bool do_scale) const
+{
+ int size = sampleSize();
+
+ unsigned val = 0;
+ if (m_Endianess == LITTLE_ENDIAN) {
+ sample = sample + size - 1;
+ for (int i = size - 1; i >= 0; --i, --sample) {
+ val = (val << 8) | (unsigned char)*sample;
+ }
+ } else {
+ for (int i = 0; i < size; ++i, ++sample) {
+ val = (val << 8) | (unsigned char)*sample;
+ }
+ }
+
+ int scale = (sizeof(unsigned) << 3) - m_SampleBits;
+ int signmask = do_scale ? (!m_IsSigned << ((sizeof(unsigned) << 3) - 1)) :
+ (-m_IsSigned << ((size << 3) - 1)) ;
+ if (do_scale) {
+ // map to int number space
+ return (val << scale) ^ signmask;
+ } else {
+ // do only sign extension
+ if (val & signmask)
+ val |= signmask;
+ return val;
+ }
+}
+
+
+void SoundFormat::convertIntToSample(int src, char *dst, bool is_scaled) const
+{
+ int size = sampleSize();
+ int scale = (sizeof(unsigned) * 8) - m_SampleBits;
+ int signmask = (!m_IsSigned << (sizeof(unsigned) * 8 - 1));
+
+ unsigned val = is_scaled ? (src ^ signmask) >> scale : src;
+ if (m_Endianess == LITTLE_ENDIAN) {
+ for (int i = 0; i < size; ++i, ++dst) {
+ (unsigned char &)*dst = val & 0xFF;
+ val >>= 8;
+ }
+ } else {
+ dst = dst - 1 + size;
+ for (int i = size - 1; i >= 0; --i, --dst) {
+ (unsigned char &)*dst = val & 0xFF;
+ val >>= 8;
+ }
+ }
+}
+
+
+void SoundFormat::convertSamplesToInts(const char *src, int *dst, size_t n, bool do_scale) const
+{
+ int size = sampleSize();
+ int scale = (sizeof(unsigned) * 8) - m_SampleBits;
+ int signmask = do_scale ? (!m_IsSigned << ((sizeof(unsigned) << 3) - 1)) :
+ (-m_IsSigned << ((size << 3) - 1)) ;
+ if (m_Endianess == LITTLE_ENDIAN) {
+ src = src - 1 + (size * n);
+ int *end = dst;
+ for (dst = dst - 1 + n; dst >= end; --dst) {
+ unsigned val = 0;
+ for (int i = size - 1; i >= 0; --i, --src) {
+ val = (val << 8) | (unsigned char)*src;
+ }
+ if (do_scale) {
+ *dst = (val << scale) ^ signmask;
+ } else if (val & signmask) {
+ *dst = val | signmask;
+ }
+ }
+ } else {
+ for (int *end = dst + n; dst < end; ++dst) {
+ unsigned val = 0;
+ for (int i = 0; i < size; ++i, ++src) {
+ val = (val << 8) | (unsigned char)*src;
+ }
+ if (do_scale) {
+ *dst = (val << scale) ^ signmask;
+ } else if (val & signmask) {
+ *dst = val | signmask;
+ }
+ }
+ }
+}
+
+
+void SoundFormat::convertIntsToSamples(const int *src, char *dst, size_t n, bool is_scaled) const
+{
+ int size = sampleSize();
+ int scale = (sizeof(unsigned) * 8) - m_SampleBits;
+ int signmask = (!m_IsSigned << (sizeof(unsigned) * 8 - 1));
+
+ if (m_Endianess == LITTLE_ENDIAN) {
+ for (const int *end = src+n; src < end; ++src) {
+ unsigned val = is_scaled ? ((unsigned)(*src ^ signmask)) >> scale : *src;
+ for (int i = 0; i < size; ++i, ++dst) {
+ (unsigned char &)*dst = val & 0xFF;
+ val >>= 8;
+ }
+ }
+ } else {
+ dst = dst - 1 + (size * n);
+ const int *end = src;
+ for (src = src - 1 + n; src >= end; --src) {
+ unsigned val = is_scaled ? ((unsigned)(*src ^ signmask)) >> scale : *src;
+ for (int i = size - 1; i >= 0; --i, --dst) {
+ (unsigned char &)*dst = val & 0xFF;
+ val >>= 8;
+ }
+ }
+ }
+}
+
+
+void SoundFormat::convertSamplesToFloat(const char *_src, float **_dst, size_t n) const
+{
+ int sample_size = sampleSize();
+ int frame_size = frameSize();
+ int scale = (sizeof(short) << 3) - m_SampleBits;
+ int signmask = !m_IsSigned << ((sizeof(short) << 3) - 1);
+ int skip = frame_size - sample_size;
+
+ if (m_Endianess == LITTLE_ENDIAN) {
+ const char *src_ch0_end = _src + frame_size * (n - 1) + sample_size - 1;
+ for (unsigned ch = 0; ch < m_Channels; ++ch) {
+ const char *src = src_ch0_end + sample_size * ch;
+ float *dst = _dst[ch];
+ float *end = dst;
+ for (dst = dst - 1 + n; dst >= end; --dst) {
+ unsigned val = 0;
+ for (int i = sample_size - 1; i >= 0; --i, --src) {
+ val = (val << 8) | (unsigned char)*src;
+ }
+ *dst = (float)(signed short)((val << scale) ^ signmask) / 32768.0f;
+ src = src - skip;
+ }
+ }
+ } else {
+ for (unsigned ch = 0; ch < m_Channels; ++ch) {
+ const char *src = _src + sample_size * ch;
+ float *dst = _dst[ch];
+ for (float *end = dst + n; dst < end; ++dst) {
+ unsigned val = 0;
+ for (int i = 0; i < sample_size; ++i, ++src) {
+ val = (val << 8) | (unsigned char)*src;
+ }
+ *dst = (float)(signed short)((val << scale) ^ signmask) / 32768.0f;
+ src = src + skip;
+ }
+ }
+ }
+}
+
+
+
+void SoundFormat::convertFloatsToSamples(const float **_src, char *_dst, size_t n) const
+{
+ int sample_size = sampleSize();
+ int frame_size = frameSize();
+ int scale = (sizeof(short) << 3) - m_SampleBits;
+ int signmask = (!m_IsSigned << (sizeof(short) << 3) - 1);
+ int skip = frame_size - sample_size;
+
+ if (m_Endianess == LITTLE_ENDIAN) {
+ for (unsigned ch = 0; ch < m_Channels; ++ch) {
+ const float *src = _src[ch];
+ char *dst = _dst + ch * sample_size;
+ for (const float *end = src+n; src < end; ++src) {
+ unsigned val = (( ((unsigned)(*src * 32768.0f)) ^ signmask)) >> scale;
+ for (int i = 0; i < sample_size; ++i, ++dst) {
+ (unsigned char &)*dst = val & 0xFF;
+ val >>= 8;
+ }
+ dst = dst + skip;
+ }
+ }
+ } else {
+ char *dst_ch0_end = _dst + frame_size * (n - 1) + sample_size - 1;
+ for (unsigned ch = 0; ch < m_Channels; ++ch) {
+ char *dst = dst_ch0_end + sample_size * ch;
+ const float *src = _src[ch];
+ const float *end = src;
+ for (src = src - 1 + n; src >= end; --src) {
+ unsigned val = (( ((unsigned)(*src * 32768.0f)) ^ signmask)) >> scale;
+ for (int i = sample_size - 1; i >= 0; --i, --dst) {
+ (unsigned char &)*dst = val & 0xFF;
+ val >>= 8;
+ }
+ dst = dst - skip;
+ }
+ }
+ }
+}
+
+
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