diff --git a/daemon/src/audio/audioloop.cpp b/daemon/src/audio/audioloop.cpp
index 445060a62a1f07c24fa3b92d61e961fc337832f1..63833f7c48e1ab6d6c8f7cfe5d0f488da57f454a 100644
--- a/daemon/src/audio/audioloop.cpp
+++ b/daemon/src/audio/audioloop.cpp
@@ -37,23 +37,21 @@
#include <cstring>
#include <cassert>
-AudioLoop::AudioLoop() :_buffer (0), _size (0), _pos (0), _sampleRate (0)
+AudioLoop::AudioLoop() : buffer_(0), size_(0), pos_(0), sampleRate_(0)
{
}
AudioLoop::~AudioLoop()
{
- delete [] _buffer;
+ delete [] buffer_;
}
void
AudioLoop::getNext (SFLDataFormat* output, size_t total_samples, short volume)
{
- size_t pos = _pos;
+ size_t pos = pos_;
- assert(_size);
-
- if(_size == 0) {
+ if (size_ == 0) {
_error("AudioLoop: Error: Audio loop size is 0");
return;
}
@@ -61,26 +59,26 @@ AudioLoop::getNext (SFLDataFormat* output, size_t total_samples, short volume)
while (total_samples) {
size_t samples = total_samples;
- if (samples > (_size-pos)) {
- samples = _size-pos;
- }
+ if (samples > (size_ - pos))
+ samples = size_ - pos;
- memcpy(output, _buffer+pos, samples*sizeof (SFLDataFormat)); // short>char conversion
+ // short->char conversion
+ memcpy(output, buffer_ + pos, samples * sizeof(SFLDataFormat));
- if (volume!=100) {
- for (size_t i=0; i<samples; i++) {
- *output = (*output * volume) /100;
+ if (volume != 100) {
+ double gain = volume * 0.01;
+ for (size_t i = 0; i < samples; i++) {
+ *output *= gain;
output++;
}
- } else {
+ } else
output += samples; // this is the destination...
- }
- pos = (pos + samples) % _size;
+ pos = (pos + samples) % size_;
total_samples -= samples;
}
- _pos = pos;
+ pos_ = pos;
}
diff --git a/daemon/src/audio/audioloop.h b/daemon/src/audio/audioloop.h
index 58e8f83c211561c92e0123d33c5d418d10d6e62c..aa2916c30e043542d11ea26ad58b5feb1d10b74c 100644
--- a/daemon/src/audio/audioloop.h
+++ b/daemon/src/audio/audioloop.h
@@ -61,36 +61,36 @@ class AudioLoop
* @param nb of int16 to send
* @param volume The volume
*/
- void getNext (SFLDataFormat* output, size_t samples, short volume=100);
+ void getNext(SFLDataFormat* output, size_t samples, short volume=100);
/**
* Reset the pointer position
*/
void reset() {
- _pos = 0;
+ pos_ = 0;
}
/**
* Accessor to the size of the buffer
* @return unsigned int The size
*/
- size_t getSize() {
- return _size;
+ size_t getSize() const {
+ return size_;
}
protected:
/** The data buffer */
- SFLDataFormat* _buffer;
+ SFLDataFormat* buffer_;
/** Number of samples inside the buffer */
- size_t _size;
+ size_t size_;
/** current position, set to 0, when initialize */
- size_t _pos;
+ size_t pos_;
/** Sample rate */
- unsigned int _sampleRate;
+ unsigned int sampleRate_;
private:
diff --git a/daemon/src/audio/audiorecord.cpp b/daemon/src/audio/audiorecord.cpp
index f84c602d27deb18cd6c6f17325beeb1aa9457739..c534fb583759eb7be74264563dcc9e142a7d579a 100644
--- a/daemon/src/audio/audiorecord.cpp
+++ b/daemon/src/audio/audiorecord.cpp
@@ -73,17 +73,9 @@ AudioRecord::AudioRecord() : fp (NULL)
AudioRecord::~AudioRecord()
{
- if (mixBuffer_) {
- delete [] mixBuffer_;
- }
-
- if (micBuffer_) {
- delete [] micBuffer_;
- }
-
- if (spkBuffer_) {
- delete [] spkBuffer_;
- }
+ delete [] mixBuffer_;
+ delete [] micBuffer_;
+ delete [] spkBuffer_;
}
void AudioRecord::setSndSamplingRate (int smplRate)
@@ -96,16 +88,12 @@ int AudioRecord::getSndSamplingRate() const
return sndSmplRate_;
}
-void AudioRecord::setRecordingOption (FILE_TYPE type, SOUND_FORMAT format, int sndSmplRate, std::string path)
+void AudioRecord::setRecordingOption(FILE_TYPE type, int sndSmplRate, const std::string &path)
{
-
fileType_ = type;
- sndFormat_ = format;
channels_ = 1;
sndSmplRate_ = sndSmplRate;
-
savePath_ = path + "/";
-
}
@@ -309,11 +297,6 @@ bool AudioRecord::setRawFile()
return false;
}
- if (sndFormat_ != INT16) { // TODO need to change INT16 to SINT16
- sndFormat_ = INT16;
- _debug ("AudioRecord::setRawFile() : using 16-bit signed integer data format for file.");
- }
-
_debug ("AudioRecord:setRawFile() : created RAW file.");
return true;
@@ -342,9 +325,7 @@ bool AudioRecord::setWavFile()
hdr.num_chans = channels_;
- if (sndFormat_ == INT16) { // TODO need to write INT16 to SINT16
- hdr.bits_per_samp = 16;
- }
+ hdr.bits_per_samp = 16;
hdr.bytes_per_samp = (SINT16) (channels_ * hdr.bits_per_samp / 8);
@@ -499,15 +480,11 @@ void AudioRecord::recData (SFLDataFormat* buffer, int nSamples)
return;
}
-
-
- if (sndFormat_ == INT16) { // TODO change INT16 to SINT16
- if (fwrite (buffer, sizeof (SFLDataFormat), nSamples, fp) != (unsigned int) nSamples)
- _warn ("AudioRecord: Could not record data! ");
- else {
- fflush (fp);
- byteCounter_ += (unsigned long) (nSamples*sizeof (SFLDataFormat));
- }
+ if (fwrite (buffer, sizeof (SFLDataFormat), nSamples, fp) != (unsigned int) nSamples)
+ _warn ("AudioRecord: Could not record data! ");
+ else {
+ fflush (fp);
+ byteCounter_ += (unsigned long) (nSamples*sizeof (SFLDataFormat));
}
}
@@ -527,25 +504,16 @@ void AudioRecord::recData (SFLDataFormat* buffer_1, SFLDataFormat* buffer_2, int
return;
}
+ for (int k = 0; k < nSamples_1; k++) {
+ mixBuffer_[k] = (buffer_1[k]+buffer_2[k]);
- if (sndFormat_ == INT16) { // TODO change INT16 to SINT16
- for (int k=0; k<nSamples_1; k++) {
-
- mixBuffer_[k] = (buffer_1[k]+buffer_2[k]);
-
-
- if (fwrite (&mixBuffer_[k], 2, 1, fp) != 1)
- _warn ("AudioRecord: Could not record data!");
- else {
- fflush (fp);
- }
- }
+ if (fwrite (&mixBuffer_[k], 2, 1, fp) != 1)
+ _warn ("AudioRecord: Could not record data!");
+ else
+ fflush (fp);
}
-
- byteCounter_ += (unsigned long) (nSamples_1*sizeof (SFLDataFormat));
-
+ byteCounter_ += (unsigned long) (nSamples_1 * sizeof(SFLDataFormat));
}
-
return;
}
diff --git a/daemon/src/audio/audiorecord.h b/daemon/src/audio/audiorecord.h
index 752ce5e3972de40caee836cbeedbd7c4de6446cf..bd7ee523aaf491c276d09f5f1669df6f1a7cc0b6 100644
--- a/daemon/src/audio/audiorecord.h
+++ b/daemon/src/audio/audiorecord.h
@@ -40,25 +40,19 @@ class AudioRecord
{
public:
+ enum FILE_TYPE { FILE_RAW, FILE_WAV };
AudioRecord();
~AudioRecord();
- /**
- * Set the sampling rate for this recorder
- */
- void setSndSamplingRate (int smplRate);
-
- /**
- * Get the recrding sampling rate
- */
+ void setSndSamplingRate(int smplRate);
+ /**
+ * Get the recrding sampling rate
+ */
int getSndSamplingRate(void) const;
- /**
- * Set the recording option
- */
- void setRecordingOption (FILE_TYPE type, SOUND_FORMAT format, int sndSmplRate, std::string path);
+ void setRecordingOption(FILE_TYPE type, int sndSmplRate, const std::string &path);
/**
* Init recording file path
@@ -184,11 +178,6 @@ class AudioRecord
*/
FILE_TYPE fileType_;
- /**
- * Sound format (SINT16/SINT32)
- */
- SOUND_FORMAT sndFormat_;
-
/**
* Number of channels
*/
diff --git a/daemon/src/audio/audiortp/AudioZrtpSession.cpp b/daemon/src/audio/audiortp/AudioZrtpSession.cpp
index 0a7ef0fd6bf2982e0d44c3c71be273b9e923d326..798e4cc53527947b0abd4506c2e537941498538a 100644
--- a/daemon/src/audio/audiortp/AudioZrtpSession.cpp
+++ b/daemon/src/audio/audiortp/AudioZrtpSession.cpp
@@ -27,6 +27,8 @@
* shall include the source code for the parts of OpenSSL used as well
* as that of the covered work.
*/
+
+#include "config.h"
#include "AudioZrtpSession.h"
#include "ZrtpSessionCallback.h"
@@ -49,14 +51,13 @@ namespace sfl
{
AudioZrtpSession::AudioZrtpSession (SIPCall * sipcall, const std::string& zidFilename) :
- // ost::SymmetricZRTPSession (ost::InetHostAddress (sipcall->getLocalIp().c_str()), sipcall->getLocalAudioPort()),
- AudioRtpSession(sipcall, Zrtp, static_cast<ost::RTPDataQueue *>(this), static_cast<ost::Thread *>(this))
- ,ost::TRTPSessionBase<ost::SymmetricRTPChannel, ost::SymmetricRTPChannel, ost::ZrtpQueue> (ost::InetHostAddress (sipcall->getLocalIp().c_str()),
- sipcall->getLocalAudioPort(),
- 0,
- ost::MembershipBookkeeping::defaultMembersHashSize,
- ost::defaultApplication())
- , _zidFilename (zidFilename)
+ AudioRtpSession(sipcall, Zrtp, this, this),
+ ost::TRTPSessionBase<ost::SymmetricRTPChannel, ost::SymmetricRTPChannel, ost::ZrtpQueue>(ost::InetHostAddress(sipcall->getLocalIp().c_str()),
+ sipcall->getLocalAudioPort(),
+ 0,
+ ost::MembershipBookkeeping::defaultMembersHashSize,
+ ost::defaultApplication()),
+ _zidFilename (zidFilename)
{
_debug ("AudioZrtpSession initialized");
initializeZid();
@@ -96,7 +97,7 @@ void AudioZrtpSession::initializeZid (void)
// xdg_config = std::string (HOMEDIR) + DIR_SEPARATOR_STR + ".cache/sflphone";
- std::string xdg_config = std::string (HOMEDIR) + DIR_SEPARATOR_STR + ".cache" + DIR_SEPARATOR_STR + PROGDIR + "/" + _zidFilename;
+ std::string xdg_config = std::string(HOMEDIR) + DIR_SEPARATOR_STR + ".cache" + DIR_SEPARATOR_STR + PACKAGE + "/" + _zidFilename;
_debug (" xdg_config %s", xdg_config.c_str());
diff --git a/daemon/src/audio/codecs/audiocodecfactory.cpp b/daemon/src/audio/codecs/audiocodecfactory.cpp
index 60191b19cce367776b41cade72f016bb9e857c7a..632fbc1f148f3ab321aea5a3a0132dbcf1e56ac8 100644
--- a/daemon/src/audio/codecs/audiocodecfactory.cpp
+++ b/daemon/src/audio/codecs/audiocodecfactory.cpp
@@ -31,6 +31,7 @@
* as that of the covered work.
*/
+#include "config.h"
#include "audiocodecfactory.h"
#include <cstdlib>
#include <algorithm> // for std::find
@@ -142,7 +143,7 @@ std::vector<sfl::Codec*> AudioCodecFactory::scanCodecDirectory()
std::vector<sfl::Codec*> codecs;
std::vector<std::string> dirToScan;
- dirToScan.push_back(std::string(HOMEDIR) + DIR_SEPARATOR_STR "." PROGDIR "/");
+ dirToScan.push_back(std::string(HOMEDIR) + DIR_SEPARATOR_STR "." PACKAGE "/");
dirToScan.push_back(CODECS_DIR "/");
const char *envDir = getenv("CODECS_PATH");
if (envDir)
@@ -162,7 +163,7 @@ std::vector<sfl::Codec*> AudioCodecFactory::scanCodecDirectory()
dirent *dirStruct;
while ( (dirStruct = readdir (dir))) {
std::string file = dirStruct->d_name ;
- if (file == CURRENT_DIR or file == PARENT_DIR)
+ if (file == "." or file == "..")
continue;
if (seemsValid (file) && !alreadyInCache (file)) {
@@ -243,8 +244,9 @@ bool AudioCodecFactory::seemsValid (const std::string &lib)
{
// The name of the shared library seems valid <==> it looks like libcodec_xxx.so
// We check this
- std::string prefix = SFL_CODEC_VALID_PREFIX;
- std::string suffix = SFL_CODEC_VALID_EXTEN;
+
+ static const std::string prefix("libcodec_");
+ static const std::string suffix(".so");
ssize_t len = lib.length() - prefix.length() - suffix.length();
if (len < 0)
@@ -257,17 +259,17 @@ bool AudioCodecFactory::seemsValid (const std::string &lib)
#ifndef HAVE_SPEEX_CODEC
- if (lib.substr (prefix.length() , len) == SPEEX_STRING_DESCRIPTION)
+ if (lib.substr (prefix.length() , len) == "speex")
return false;
#endif
#ifndef HAVE_GSM_CODEC
- if (lib.substr (prefix.length() , len) == GSM_STRING_DESCRIPTION)
+ if (lib.substr (prefix.length() , len) == "gsm")
return false;
#endif
#ifndef BUILD_ILBC
- if (lib.substr (prefix.length() , len) == ILBC_STRING_DESCRIPTION)
+ if (lib.substr (prefix.length() , len) == "ilbc")
return false;
#endif
diff --git a/daemon/src/audio/recordable.cpp b/daemon/src/audio/recordable.cpp
index e98a39ba32511be7a98a6b7494d73caf99415bc8..2ee5ffe76bd78bf51275abbec528ae2ba4bb6462 100644
--- a/daemon/src/audio/recordable.cpp
+++ b/daemon/src/audio/recordable.cpp
@@ -32,13 +32,8 @@
Recordable::Recordable() : recorder (&recAudio, Manager::instance().getMainBuffer())
{
-
_debug("-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Recordable Constructor -=-=-=-=-=-=-=-=-=--=-=-=-");
-
- FILE_TYPE fileType = FILE_WAV;
- SOUND_FORMAT soundFormat = INT16;
-
- recAudio.setRecordingOption (fileType, soundFormat, 8000, Manager::instance().audioPreference.getRecordpath());
+ recAudio.setRecordingOption (AudioRecord::FILE_WAV, 8000, Manager::instance().audioPreference.getRecordpath());
}
diff --git a/daemon/src/audio/samplerateconverter.cpp b/daemon/src/audio/samplerateconverter.cpp
index 8bfcbeb68756a7c16f0aec2c85ea6467a9692c3c..f2654c5d30c08a23ea4f458739db04fadd584381 100644
--- a/daemon/src/audio/samplerateconverter.cpp
+++ b/daemon/src/audio/samplerateconverter.cpp
@@ -39,8 +39,8 @@ SamplerateConverter::SamplerateConverter (int freq) : _maxFreq(freq)
_samples = (freq * 20) / 1000; // start with 20 ms buffers
- _floatBufferIn = new float32[_samples];
- _floatBufferOut = new float32[_samples];
+ _floatBufferIn = new float[_samples];
+ _floatBufferOut = new float[_samples];
}
SamplerateConverter::~SamplerateConverter (void)
@@ -61,7 +61,6 @@ SamplerateConverter::Short2FloatArray (const short *in, float *out, int len)
out[len] = (float) in[len] * .000030517578125f;
}
-//TODO Add ifdef for int16 or float32 type
void SamplerateConverter::resample (SFLDataFormat* dataIn , SFLDataFormat* dataOut , int inputFreq , int outputFreq , int nbSamples)
{
double sampleFactor = (double) outputFreq / inputFreq;
@@ -77,8 +76,8 @@ void SamplerateConverter::resample (SFLDataFormat* dataIn , SFLDataFormat* dataO
_samples = maxSamples;
delete [] _floatBufferIn;
delete [] _floatBufferOut;
- _floatBufferIn = new float32[_samples];
- _floatBufferOut = new float32[_samples];
+ _floatBufferIn = new float[_samples];
+ _floatBufferOut = new float[_samples];
}
SRC_DATA src_data;
diff --git a/daemon/src/audio/samplerateconverter.h b/daemon/src/audio/samplerateconverter.h
index 6d17b8d671f84e056ec32cdc1a0c1a5f9fd17af2..58dbd34409483a6a4dc049514d6d516ed8c036a9 100644
--- a/daemon/src/audio/samplerateconverter.h
+++ b/daemon/src/audio/samplerateconverter.h
@@ -78,8 +78,8 @@ class SamplerateConverter
SamplerateConverter& operator= (const SamplerateConverter& rh);
/* temporary buffers */
- float32* _floatBufferIn;
- float32* _floatBufferOut;
+ float * _floatBufferIn;
+ float * _floatBufferOut;
size_t _samples; // size in samples of temporary buffers
int _maxFreq; // maximal output frequency
diff --git a/daemon/src/audio/sound/audiofile.cpp b/daemon/src/audio/sound/audiofile.cpp
index c744aca12eb7e06cbce7c5f2a1a0a8115ebc3920..ece603e36db9a088f23b6b2d60b6ee716e7b6e9d 100644
--- a/daemon/src/audio/sound/audiofile.cpp
+++ b/daemon/src/audio/sound/audiofile.cpp
@@ -73,51 +73,40 @@ RawFile::RawFile(const std::string& name, sfl::AudioCodec* codec, unsigned int s
SFLDataFormat *monoBuffer = new SFLDataFormat[decodedSize];
SFLDataFormat *bufpos = monoBuffer;
unsigned char *filepos = reinterpret_cast<unsigned char *>(fileBuffer);
- _size = decodedSize;
- while(length >= encFrameSize) {
+ size_ = decodedSize;
+ while (length >= encFrameSize) {
bufpos += audioCodec->decode (bufpos, filepos, encFrameSize);
filepos += encFrameSize;
length -= encFrameSize;
}
- delete[] fileBuffer;
-
- if (sampleRate == audioRate) {
-#ifdef DATAFORMAT_IS_FLOAT
- _buffer = new SFLDataFormat[_size];
- src_short_to_float_array (monoBuffer, _buffer, _size);
- delete[] monoBuffer;
-#else
- _buffer = monoBuffer;
-#endif
-
- } else {
+ delete [] fileBuffer;
+
+ if (sampleRate == audioRate)
+ buffer_ = monoBuffer;
+ else {
double factord = (double) sampleRate / audioRate;
- float* floatBufferIn = new float[_size];
- int sizeOut = ceil(factord*_size);
- src_short_to_float_array (monoBuffer, floatBufferIn, _size);
- delete[] monoBuffer;
- SFLDataFormat* _buffer = new SFLDataFormat[sizeOut];
+ float* floatBufferIn = new float[size_];
+ int sizeOut = ceil(factord * size_);
+ src_short_to_float_array (monoBuffer, floatBufferIn, size_);
+ delete [] monoBuffer;
+ delete [] buffer_;
+ buffer_ = new SFLDataFormat[sizeOut];
SRC_DATA src_data;
src_data.data_in = floatBufferIn;
- src_data.input_frames = _size;
+ src_data.input_frames = size_;
src_data.output_frames = sizeOut;
src_data.src_ratio = factord;
-#ifdef DATAFORMAT_IS_FLOAT
- src_data.data_out = _buffer;
- src_simple (&src_data, SRC_SINC_BEST_QUALITY, 1);
-#else
float* floatBufferOut = new float[sizeOut];
src_data.data_out = floatBufferOut;
src_simple (&src_data, SRC_SINC_BEST_QUALITY, 1);
- src_float_to_short_array (floatBufferOut, _buffer, src_data.output_frames_gen);
+ src_float_to_short_array (floatBufferOut, buffer_, src_data.output_frames_gen);
delete [] floatBufferOut;
-#endif
delete [] floatBufferIn;
- _size = src_data.output_frames_gen;
+ size_ = src_data.output_frames_gen;
}
}
@@ -148,8 +137,8 @@ WaveFile::WaveFile (const std::string& fileName, unsigned int audioSamplingRate)
SINT32 chunk_size; // fmt chunk size
unsigned short formatTag; // data compression tag
- fileStream.read ( (char*) &chunk_size, 4); // Read fmt chunk size.
- fileStream.read ( (char*) &formatTag, 2);
+ fileStream.read(reinterpret_cast<char*>(&chunk_size), 4); // Read fmt chunk size.
+ fileStream.read(reinterpret_cast<char*>(&formatTag), 2);
if (formatTag != 1) // PCM = 1, FLOAT = 3
throw AudioFileException("File contains an unsupported data format type");
@@ -158,12 +147,12 @@ WaveFile::WaveFile (const std::string& fileName, unsigned int audioSamplingRate)
SINT16 chan;
fileStream.read ( (char*) &chan, 2);
- if(chan > 2)
+ if (chan > 2)
throw AudioFileException("WaveFile: unsupported number of channels");
// Get file sample rate from the header.
SINT32 srate;
- fileStream.read ( (char*) &srate, 4);
+ fileStream.read( (char*) &srate, 4);
SINT32 avgb;
fileStream.read ( (char*) &avgb, 4);
@@ -185,7 +174,7 @@ WaveFile::WaveFile (const std::string& fileName, unsigned int audioSamplingRate)
// Sample rate converter initialized with 88200 sample long
int converterSamples = ((unsigned int)srate > audioSamplingRate) ? srate : audioSamplingRate;
- SamplerateConverter _converter (converterSamples);
+ SamplerateConverter converter_(converterSamples);
// Get length of data from the header.
SINT32 bytes;
@@ -197,31 +186,28 @@ WaveFile::WaveFile (const std::string& fileName, unsigned int audioSamplingRate)
nbSamples, bytes, blockal, srate, avgb, chunk_size, dt);
// Should not be longer than a minute
- if (nbSamples > (unsigned int) (60*srate))
- nbSamples = 60*srate;
+ if (nbSamples > static_cast<unsigned int>(60 * srate))
+ nbSamples = 60 * srate;
SFLDataFormat *tempBuffer = new SFLDataFormat[nbSamples];
- if (!tempBuffer)
- throw AudioFileException("Could not allocate temporary buffer");
- fileStream.read ( (char *) tempBuffer, nbSamples*sizeof (SFLDataFormat));
+ fileStream.read (reinterpret_cast<char *>(tempBuffer), nbSamples * sizeof(SFLDataFormat));
// mix two channels together if stereo
- if(chan == 2) {
- for(unsigned int i = 0; i < nbSamples-1; i+=2)
- tempBuffer[i/2] = (tempBuffer[i] + tempBuffer[i+1]) / 2;
- nbSamples /= 2;
+ if (chan == 2) {
+ for (unsigned int i = 0; i < nbSamples - 1; i += 2)
+ tempBuffer[static_cast<size_t>(i * 0.5)] = (tempBuffer[i] + tempBuffer[i + 1]) * 0.5;
+ nbSamples *= 0.5;
}
- if ((unsigned int)srate != audioSamplingRate) {
+ if ((unsigned int) srate != audioSamplingRate) {
int outSamples = ((float) nbSamples * ( (float) audioSamplingRate / (float) srate));
- _buffer = new SFLDataFormat[outSamples];
- _converter.resample (tempBuffer, _buffer, srate, audioSamplingRate, nbSamples);
- delete[] tempBuffer;
- } else {
- _buffer = tempBuffer;
- }
-
- _size = nbSamples;
- _sampleRate = audioSamplingRate;
+ buffer_ = new SFLDataFormat[outSamples];
+ converter_.resample (tempBuffer, buffer_, srate, audioSamplingRate, nbSamples);
+ delete [] tempBuffer;
+ } else
+ buffer_ = tempBuffer;
+
+ size_ = nbSamples;
+ sampleRate_ = audioSamplingRate;
}
diff --git a/daemon/src/audio/sound/tone.cpp b/daemon/src/audio/sound/tone.cpp
index d45064ef43a9d65b921a813601c1af642cfdd094..a19e562b3437802b7c5b49e5f9ce58ffcb55672f 100644
--- a/daemon/src/audio/sound/tone.cpp
+++ b/daemon/src/audio/sound/tone.cpp
@@ -35,31 +35,26 @@
* YM: 2006-11-15: changes unsigned int to std::string::size_type, thanks to Pierre Pomes (AMD64 compilation)
*/
#include "tone.h"
-#include <math.h>
+#include <cmath>
#include <cstdlib>
#include <cstring>
-#define TABLE_LENGTH 4096
-double TWOPI = 2 * M_PI;
+static const double TWOPI = 2.0 * M_PI;
-Tone::Tone (const std::string& definition, unsigned int sampleRate) : AudioLoop(), _sampleRate (sampleRate), _xhigher (0.0), _xlower (0.0)
+Tone::Tone(const std::string& definition, unsigned int sampleRate) :
+ sampleRate_(sampleRate), xhigher_(0.0), xlower_(0.0)
{
fillWavetable();
- genBuffer (definition); // allocate memory with definition parameter
-}
-
-Tone::~Tone()
-{
+ genBuffer(definition); // allocate memory with definition parameter
}
void
-Tone::genBuffer (const std::string& definition)
+Tone::genBuffer(const std::string& definition)
{
- if (definition.empty()) {
+ if (definition.empty())
return;
- }
- _size = 0;
+ size_ = 0;
SFLDataFormat* buffer = new SFLDataFormat[SIZEBUF]; //1kb
SFLDataFormat* bufferPos = buffer;
@@ -76,67 +71,61 @@ Tone::genBuffer (const std::string& definition)
do {
posEnd = definition.find (',', posStart);
- if (posEnd == std::string::npos) {
+ if (posEnd == std::string::npos)
posEnd = deflen;
- }
+ /* begin scope */
{
// Sample string: "350+440" or "350+440/2000,244+655/2000"
int freq1, freq2, time;
- s = definition.substr (posStart, posEnd-posStart);
+ s = definition.substr(posStart, posEnd-posStart);
// The 1st frequency is before the first + or the /
- std::string::size_type pos_plus = s.find ('+');
- std::string::size_type pos_slash = s.find ('/');
- std::string::size_type len = s.length();
- std::string::size_type endfrequency = 0;
+ size_t pos_plus = s.find('+');
+ size_t pos_slash = s.find('/');
+ size_t len = s.length();
+ size_t endfrequency = 0;
if (pos_slash == std::string::npos) {
time = 0;
endfrequency = len;
} else {
- time = atoi ( (s.substr (pos_slash+1,len-pos_slash-1)).data());
+ time = atoi(s.substr(pos_slash + 1, len - pos_slash - 1).c_str());
endfrequency = pos_slash;
}
// without a plus = 1 frequency
if (pos_plus == std::string::npos) {
- freq1 = atoi ( (s.substr (0,endfrequency)).data());
+ freq1 = atoi(s.substr(0, endfrequency).c_str());
freq2 = 0;
} else {
- freq1 = atoi ( (s.substr (0,pos_plus)).data());
- freq2 = atoi ( (s.substr (pos_plus+1, endfrequency-pos_plus-1)).data());
+ freq1 = atoi(s.substr(0, pos_plus).c_str());
+ freq2 = atoi(s.substr(pos_plus + 1, endfrequency - pos_plus - 1).c_str());
}
// If there is time or if it's unlimited
- if (time == 0) {
- count = _sampleRate;
- } else {
- count = (_sampleRate * time) / 1000;
- }
+ if (time == 0)
+ count = sampleRate_;
+ else
+ count = (sampleRate_ * time) / 1000;
// Generate SAMPLING_RATE samples of sinus, buffer is the result
- _debug ("genSin(%d, %d)", freq1, freq2);
- genSin (bufferPos, freq1, freq2, count);
+ _debug("genSin(%d, %d)", freq1, freq2);
+ genSin(bufferPos, freq1, freq2, count);
// To concatenate the different buffers for each section.
- _size += (count);
+ size_ += count;
+ bufferPos += count;
+ } /* end scope */
- bufferPos += (count);
- }
-
- posStart = posEnd+1;
+ posStart = posEnd + 1;
} while (posStart < deflen);
- _buffer = new SFLDataFormat[_size];
-
- memcpy (_buffer, buffer, _size*sizeof (SFLDataFormat)); // copy char, not SFLDataFormat.
-
- delete[] buffer;
+ buffer_ = new SFLDataFormat[size_];
- buffer=0;
+ memcpy(buffer_, buffer, size_ * sizeof(SFLDataFormat)); // copy char, not SFLDataFormat.
- bufferPos=0;
+ delete [] buffer;
}
void
@@ -144,9 +133,8 @@ Tone::fillWavetable()
{
double tableSize = (double) TABLE_LENGTH;
- for (int i = 0; i < TABLE_LENGTH; i++) {
- _wavetable[i] = sin ( ( (double) i / (tableSize - 1.0)) * TWOPI);
- }
+ for (int i = 0; i < TABLE_LENGTH; ++i)
+ wavetable_[i] = sin((static_cast<double>(i) / (tableSize - 1.0)) * TWOPI);
}
double
@@ -156,24 +144,24 @@ Tone::interpolate (double x)
double yi_0, yi_1, A, B;
xi_0 = (int) x;
- xi_1 = xi_0+1;
+ xi_1 = xi_0 + 1;
- yi_0 =_wavetable[xi_0];
- yi_1 = _wavetable[xi_1];
+ yi_0 = wavetable_[xi_0];
+ yi_1 = wavetable_[xi_1];
A = (x - xi_0);
B = 1.0 - A;
- return A*yi_0 + B*yi_1;
+ return (A * yi_0) + (B * yi_1);
}
void
Tone::genSin (SFLDataFormat* buffer, int frequency1, int frequency2, int nb)
{
- _xhigher = 0.0;
- _xlower = 0.0;
+ xhigher_ = 0.0;
+ xlower_ = 0.0;
- double sr = (double) _sampleRate;
+ double sr = (double) sampleRate_;
double tableSize = (double) TABLE_LENGTH;
double N_h = sr / (double) (frequency1);
@@ -182,27 +170,25 @@ Tone::genSin (SFLDataFormat* buffer, int frequency1, int frequency2, int nb)
double dx_h = tableSize / N_h;
double dx_l = tableSize / N_l;
- double x_h = _xhigher;
- double x_l = _xlower;
+ double x_h = xhigher_;
+ double x_l = xlower_;
- double amp = (double) SFLDataAmplitude;
+ static const double DATA_AMPLITUDE = 2047;
+ double amp = DATA_AMPLITUDE;
for (int t = 0; t < nb; t ++) {
- buffer[t] = (int16) (amp* (interpolate (x_h) + interpolate (x_l)));
+ buffer[t] = static_cast<SFLDataFormat>(amp * (interpolate(x_h) + interpolate(x_l)));
x_h += dx_h;
x_l += dx_l;
- if (x_h > tableSize) {
+ while (x_h > tableSize)
x_h -= tableSize;
- }
- if (x_l > tableSize) {
+ while (x_l > tableSize)
x_l -= tableSize;
- }
}
- _xhigher = x_h;
- _xlower = x_l;
-
+ xhigher_ = x_h;
+ xlower_ = x_l;
}
diff --git a/daemon/src/audio/sound/tone.h b/daemon/src/audio/sound/tone.h
index 1cfa3348411298b0afef9b042ee6065202c06666..fc20ef90f8bb3252ac66af9f10339eb90e4a9928 100644
--- a/daemon/src/audio/sound/tone.h
+++ b/daemon/src/audio/sound/tone.h
@@ -36,8 +36,6 @@
#include <string>
#include "audio/audioloop.h"
-#define TABLE_LENGTH 4096
-
/**
* @file tone.h
* @brief Tone sample (dial, busy, ring, congestion)
@@ -52,11 +50,6 @@ class Tone : public AudioLoop
*/
Tone (const std::string& definition, unsigned int sampleRate);
- /**
- * Destructor
- */
- ~Tone();
-
/** The different kind of tones */
enum TONEID {
TONE_DIALTONE = 0,
@@ -74,17 +67,17 @@ class Tone : public AudioLoop
* @param nb are the number of int16 (mono) to generate
* by example nb=5 generate 10 int16, 5 for the left, 5 for the right
*/
- void genSin (SFLDataFormat* buffer, int frequency1, int frequency2, int nb);
+ void genSin(SFLDataFormat* buffer, int frequency1, int frequency2, int nb);
/**
*
*/
- void fillWavetable (void);
+ void fillWavetable(void);
/**
*
*/
- double interpolate (double x);
+ double interpolate(double x);
private:
@@ -93,15 +86,16 @@ class Tone : public AudioLoop
* allocate the memory with the definition
* @param definition String that contain frequency/time of the tone.
*/
- void genBuffer (const std::string& definition);
+ void genBuffer(const std::string& definition);
/** Sample rate */
- unsigned int _sampleRate;
+ unsigned int sampleRate_;
- double _wavetable[TABLE_LENGTH];
+ static const int TABLE_LENGTH = 4096;
+ double wavetable_[TABLE_LENGTH];
- double _xhigher;
- double _xlower;
+ double xhigher_;
+ double xlower_;
};
#endif // __TONE_H__
diff --git a/daemon/src/global.h b/daemon/src/global.h
index b7d8458aa872f386a33e8e552f01b4dae30033b1..ef8b085fe4dcce223cc4c254cd4007aae3149ceb 100644
--- a/daemon/src/global.h
+++ b/daemon/src/global.h
@@ -49,38 +49,13 @@
#define XDG_CACHE_HOME (getenv ("XDG_CACHE_HOME"))
const char * const ZRTP_ZID_FILENAME = "sfl.zid";
-typedef float float32;
-typedef short int16;
-
//useful typedefs.
typedef signed short SINT16;
typedef signed int SINT32;
-typedef unsigned long FILE_TYPE;
-typedef unsigned long SOUND_FORMAT;
-
-const FILE_TYPE FILE_RAW = 1;
-const FILE_TYPE FILE_WAV = 2;
-
-static const SOUND_FORMAT INT16 = 0x2; // TODO shold change these symbols
-static const SOUND_FORMAT INT32 = 0x8;
-
#define PIDFILE "sfl.pid"
-#ifdef DATAFORMAT_IS_FLOAT
-#define SFLDataFormat float32
-#define SFLDataFormatString "Float32"
-#define SFLDataAmplitude 0.05
-#else
-#define SFLDataFormat int16
-#define SFLDataFormatString "Int16"
-#define SFLDataAmplitude (32767 >> 4)
-#endif
-
-#define PROGNAME "sflphoned" /** Binary name */
-#define PROGDIR "sflphone" /** Program directory */
-#define RINGDIR "ringtones" /** Ringtones directory */
-#define CODECDIR "codecs" /** Codecs directory */
+typedef short SFLDataFormat;
#define SIZEBUF 400000 /** About 12 sec of buffering at 8000 Hz*/
@@ -92,19 +67,11 @@ static const SOUND_FORMAT INT32 = 0x8;
#define PCM_DSNOOP "plug:dsnoop" /** Alsa plugin for microphone sharing */
#define PCM_DMIX_DSNOOP "dmix/dsnoop" /** Audio profile using Alsa dmix/dsnoop */
-#define SFL_CODEC_VALID_PREFIX "libcodec_" /** Valid prefix for codecs shared library */
-#define SFL_CODEC_VALID_EXTEN ".so" /** Valid extension for codecs shared library */
-#define CURRENT_DIR "." /** Current directory */
-#define PARENT_DIR ".." /** Parent directory */
-
#define SFL_PCM_BOTH 0x0021 /** To open both playback and capture devices */
#define SFL_PCM_PLAYBACK 0x0022 /** To open playback device only */
#define SFL_PCM_CAPTURE 0x0023 /** To open capture device only */
-#define SFL_PCM_RINGTONE 0x0024
+#define SFL_PCM_RINGTONE 0x0024
-#define GSM_STRING_DESCRIPTION "gsm" /** GSM codec string description */
-#define SPEEX_STRING_DESCRIPTION "speex" /** SPEEX codec string description */
-#define ILBC_STRING_DESCRIPTION "ilbc" /** Ilbc codec string description */
#define RINGTONE_ENABLED TRUE_STR /** Custom ringtone enable or not */
#define DISPLAY_DIALPAD TRUE_STR /** Display dialpad or not */
#define DISPLAY_VOLUME_CONTROLS TRUE_STR /** Display the volume controls or not */
@@ -126,9 +93,6 @@ static const SOUND_FORMAT INT32 = 0x8;
#define DEFAULT_SIP_PORT 5060
#define DEFAULT_SIP_TLS_PORT 5061
-#define HOOK_DEFAULT_SIP_FIELD "X-sflphone-url"
-#define HOOK_DEFAULT_URL_COMMAND "x-www-browser"
-
/** Enumeration that contains known audio payloads */
enum {
// http://www.iana.org/assignments/rtp-parameters
diff --git a/daemon/src/managerimpl.cpp b/daemon/src/managerimpl.cpp
index a59b99b4a88ae5c7755b433ce818d286251585f1..7ce6d1dc8447010503928efd10ec4fbcf9cc7d70 100644
--- a/daemon/src/managerimpl.cpp
+++ b/daemon/src/managerimpl.cpp
@@ -1743,6 +1743,7 @@ void ManagerImpl::ringtone (const std::string& accountID)
std::string ringchoice = account->getRingtonePath();
if (ringchoice.find (DIR_SEPARATOR_STR) == std::string::npos) {
// check inside global share directory
+ static const char * RINGDIR = "ringtones";
ringchoice = std::string (PROGSHAREDIR) + DIR_SEPARATOR_STR
+ RINGDIR + DIR_SEPARATOR_STR + ringchoice;
}
@@ -1819,7 +1820,7 @@ ManagerImpl::getTelephoneFile ()
std::string ManagerImpl::getConfigFile (void) const
{
std::string configdir = std::string (HOMEDIR) + DIR_SEPARATOR_STR + ".config"
- + DIR_SEPARATOR_STR + PROGDIR;
+ + DIR_SEPARATOR_STR + PACKAGE;
if (XDG_CONFIG_HOME != NULL) {
std::string xdg_env = std::string (XDG_CONFIG_HOME);
@@ -1833,6 +1834,7 @@ std::string ManagerImpl::getConfigFile (void) const
_debug ("Cannot create directory: %m");
}
+ static const char * PROGNAME = "sflphoned";
return configdir + DIR_SEPARATOR_STR + PROGNAME + ".yml";
}