diff --git a/sflphone-common/src/audio/alsa/alsalayer.cpp b/sflphone-common/src/audio/alsa/alsalayer.cpp
index 5c19f3da8cce792c8b86742700bcfe1aceffd651..1914560d0e04df03be8d9d9917b5ee163944b273 100644
--- a/sflphone-common/src/audio/alsa/alsalayer.cpp
+++ b/sflphone-common/src/audio/alsa/alsalayer.cpp
@@ -902,7 +902,7 @@ AlsaLayer::soundCardGetIndex (std::string description)
void AlsaLayer::audioCallback(void)
{
- int toGet, urgentAvailBytes, normalAvailBytes, maxBytes;
+ int toGet, urgentAvailBytes, normalAvailBytes;
unsigned short spkrVolume, micVolume;
AudioLoop *tone;
AudioLoop *file_tone;
diff --git a/sflphone-common/src/audio/audiorecord.cpp b/sflphone-common/src/audio/audiorecord.cpp
index 06d97d0b30d0ee58919e957e3f243245a880ae1e..491996185d70713f17304630842f19358789f995 100644
--- a/sflphone-common/src/audio/audiorecord.cpp
+++ b/sflphone-common/src/audio/audiorecord.cpp
@@ -494,7 +494,7 @@ void AudioRecord::recData (SFLDataFormat* buffer, int nSamples)
}
-void AudioRecord::recData (SFLDataFormat* buffer_1, SFLDataFormat* buffer_2, int nSamples_1, int nSamples_2)
+void AudioRecord::recData (SFLDataFormat* buffer_1, SFLDataFormat* buffer_2, int nSamples_1, int nSamples_2 UNUSED)
{
if (recordingEnabled_) {
diff --git a/sflphone-common/src/audio/audiorecorder.cpp b/sflphone-common/src/audio/audiorecorder.cpp
index bb846e022b3a728c3bca022514583d0e84e30663..b12a3016e92b7de9cdf37a3e940059782e7a11ce 100644
--- a/sflphone-common/src/audio/audiorecorder.cpp
+++ b/sflphone-common/src/audio/audiorecorder.cpp
@@ -70,10 +70,6 @@ void AudioRecorder::run (void)
if(availBytes > 0) {
- int got = mbuffer->getData(buffer, availBytes, 100, recorderId);
-
- int availBytesAfter = mbuffer->availForGet(recorderId);
-
arecord->recData(buffer, availBytes/sizeof(SFLDataFormat));
}
diff --git a/sflphone-common/src/audio/dcblocker.cpp b/sflphone-common/src/audio/dcblocker.cpp
index b1a18eb1a2e6ddcf97568825cd057df55a8b42fc..06505b428207bc3c25639497e1f06f78ef2285bb 100644
--- a/sflphone-common/src/audio/dcblocker.cpp
+++ b/sflphone-common/src/audio/dcblocker.cpp
@@ -42,9 +42,9 @@ void DcBlocker::reset()
_ym1 = 0;
}
-void DcBlocker::putData(SFLDataFormat *inputData, int nbBytes) {}
+void DcBlocker::putData(SFLDataFormat *inputData UNUSED, int nbBytes UNUSED) {}
-int DcBlocker::getData(SFLDataFormat *outputData) { return 0; }
+int DcBlocker::getData(SFLDataFormat *outputData UNUSED) { return 0; }
void DcBlocker::process (SFLDataFormat *data, int nbBytes)
{
@@ -81,4 +81,4 @@ int DcBlocker::process(SFLDataFormat *inputData, SFLDataFormat *outputData, int
}
-void DcBlocker::process(SFLDataFormat *micData, SFLDataFormat *spkrData, SFLDataFormat *outputData, int nbBytes) {}
+void DcBlocker::process(SFLDataFormat *micData UNUSED, SFLDataFormat *spkrData UNUSED, SFLDataFormat *outputData UNUSED, int nbBytes UNUSED) {}
diff --git a/sflphone-common/src/audio/delaydetection.cpp b/sflphone-common/src/audio/delaydetection.cpp
index 5c6ab4d742644ed469eb6ab426030680173ab82b..16326c12e9070f04b9ccf2c3d1d48de966a2bef3 100644
--- a/sflphone-common/src/audio/delaydetection.cpp
+++ b/sflphone-common/src/audio/delaydetection.cpp
@@ -32,7 +32,6 @@
#include "delaydetection.h"
#include "math.h"
-// #include <stdio.h>
#include <string.h>
#include <samplerate.h>
@@ -163,7 +162,7 @@ void DelayDetection::putData(SFLDataFormat *inputData, int nbBytes)
}
-int DelayDetection::getData(SFLDataFormat *outputData) { return 0; }
+int DelayDetection::getData(SFLDataFormat *outputData UNUSED) { return 0; }
void DelayDetection::process(SFLDataFormat *inputData, int nbBytes) {
@@ -184,19 +183,6 @@ void DelayDetection::process(SFLDataFormat *inputData, int nbBytes) {
downsampleData(tmp, down, nbSamples, _downsamplingFactor);
- /*
- for(int i = 0; i < 10; i++)
- _debug("up: %.10f", tmp[i]);
-
- for(int i = 0; i < 10; i++)
- _debug("down: %.10f", down[i]);
-
- bandpassFilter(down, nbSamples/_downsamplingFactor);
-
- for(int i = 0; i < 10; i++)
- _debug("band: %.10f", down[i]);
- */
-
memcpy(_captureDataDown+(_nbMicSampleStored/_downsamplingFactor), down, (nbSamples/_downsamplingFactor)*sizeof(float));
_nbMicSampleStored += nbSamples;
@@ -208,50 +194,29 @@ void DelayDetection::process(SFLDataFormat *inputData, int nbBytes) {
else
return;
- /*
- for(int i = 0; i < 10; i++)
- _debug("spkrRef: %.10f", _spkrReferenceDown[i]);
-
- for(int i = 0; i < 10; i++)
- _debug("micSeg: %.10f", _captureDataDown[i]);
- */
-
_debug("_spkrDownSize: %d, _micDownSize: %d", _spkrDownSize, _micDownSize);
crossCorrelate(_spkrReferenceDown, _captureDataDown, _correlationResult, _micDownSize, _spkrDownSize);
int maxIndex = getMaxIndex(_correlationResult, _spkrDownSize);
_debug("MaxIndex: %d", maxIndex);
-
- // reset();
}
-int DelayDetection::process(SFLDataFormat *intputData, SFLDataFormat *outputData, int nbBytes) { return 0; }
+int DelayDetection::process(SFLDataFormat *intputData UNUSED, SFLDataFormat *outputData UNUSED, int nbBytes UNUSED) { return 0; }
-void DelayDetection::process(SFLDataFormat *micData, SFLDataFormat *spkrData, SFLDataFormat *outputData, int nbBytes) {}
+void DelayDetection::process(SFLDataFormat *micData UNUSED, SFLDataFormat *spkrData UNUSED, SFLDataFormat *outputData UNUSED, int nbBytes UNUSED) {}
void DelayDetection::crossCorrelate(float *ref, float *seg, float *res, int refSize, int segSize) {
_debug("CrossCorrelate");
- int counter = 0;
-
// Output has same size as the
int rsize = refSize;
int ssize = segSize;
int tmpsize = segSize-refSize+1;
- /*
- for(int i = 0; i < 32; i++)
- _debug("ref: %.10f", ref[i]);
-
- for(int i = 0; i < 150; i++)
- _debug("seg: %.10f", seg[i]);
- */
-
// perform autocorrelation on reference signal
float acref = correlate(ref, ref, rsize);
- // _debug("acref: %f", acref);
// perform crossrelation on signal
float acseg = 0.0;
@@ -259,7 +224,6 @@ void DelayDetection::crossCorrelate(float *ref, float *seg, float *res, int refS
while(--tmpsize) {
--ssize;
acseg = correlate(seg+tmpsize, seg+tmpsize, rsize);
- // _debug("acseg: %f", acseg);
res[ssize] = correlate(ref, seg+tmpsize, rsize);
r = sqrt(acref*acseg);
@@ -273,7 +237,6 @@ void DelayDetection::crossCorrelate(float *ref, float *seg, float *res, int refS
int i = 0;
while(rsize) {
acseg = correlate(seg, seg, rsize);
- // _debug("acseg: %f", acseg);
res[ssize-1] = correlate(ref+i, seg, rsize);
r = sqrt(acref*acseg);
@@ -302,8 +265,6 @@ double DelayDetection::correlate(float *sig1, float *sig2, short size) {
void DelayDetection::convertInt16ToFloat32(SFLDataFormat *input, float *output, int nbSamples) {
- // factor is 1/(2^15), used to rescale the short int range to the
- // [-1.0 - 1.0] float range.
#define S2F_FACTOR .000030517578125f;
int len = nbSamples;
@@ -316,26 +277,6 @@ void DelayDetection::convertInt16ToFloat32(SFLDataFormat *input, float *output,
void DelayDetection::downsampleData(float *input, float *output, int nbSamples, int factor) {
- /*
- float tmp[nbSamples];
-
- for(int i = 0; i < nbSamples; i++) {
- tmp[i] = _decimationFilter.getOutputSample(input[i]);
- }
-
- int j;
- for(j=_remainingIndex; j<nbSamples; j+=factor) {
- output[j] = tmp[j];
- }
- _remainingIndex = j - nbSamples;
- */
-
- /*
- double downsampleFactor = (double) samplerate1 / samplerate2;
-
- int nbSamplesMax = (int) (samplerate1 * getFramesize() / 1000);
- */
-
int _src_err;
SRC_STATE *_src_state = src_new (SRC_LINEAR, 1, &_src_err);
@@ -351,13 +292,7 @@ void DelayDetection::downsampleData(float *input, float *output, int nbSamples,
src_data.src_ratio = downfactor;
src_data.end_of_input = 0; // More data will come
- //src_short_to_float_array (dataIn, _floatBufferUpMic, nbSamples);
- //_debug("downsample %d %f %d" , src_data.output_frames, src_data.src_ratio , nbSamples);
src_process (_src_state, &src_data);
- //_debug("downsample %d %f %d" , src_data.output_frames, src_data.src_ratio , nbSamples);
- // nbSamples = (src_data.output_frames_gen > nbSamplesMax) ? nbSamplesMax : src_data.output_frames_gen;
- //_debug("downsample %d %f %d" , src_data.output_frames, src_data.src_ratio , nbSamples);
- // src_float_to_short_array (_floatBufferDownMic , dataOut , nbSamples);
}
}
diff --git a/sflphone-common/src/audio/echocancel.cpp b/sflphone-common/src/audio/echocancel.cpp
index aab1e8f9989fc18c7c79315ef6dd64c76de12018..612b1dbed76796b44102813c77eb7450ae5bf8c6 100644
--- a/sflphone-common/src/audio/echocancel.cpp
+++ b/sflphone-common/src/audio/echocancel.cpp
@@ -241,7 +241,7 @@ int EchoCancel::getData(SFLDataFormat *outputData)
return copied;
}
-void EchoCancel::process(SFLDataFormat *data, int nbBytes) {}
+void EchoCancel::process(SFLDataFormat *data UNUSED, int nbBytes UNUSED) {}
int EchoCancel::process(SFLDataFormat *inputData, SFLDataFormat *outputData, int nbBytes)
@@ -330,7 +330,7 @@ int EchoCancel::process(SFLDataFormat *inputData, SFLDataFormat *outputData, int
return nbFrame * _smplPerFrame;
}
-void EchoCancel::process(SFLDataFormat *micData, SFLDataFormat *spkrData, SFLDataFormat *outputData, int nbBytes){
+void EchoCancel::process(SFLDataFormat *micData UNUSED, SFLDataFormat *spkrData UNUSED, SFLDataFormat *outputData UNUSED, int nbBytes UNUSED){
}
diff --git a/sflphone-common/src/audio/echocancel.h b/sflphone-common/src/audio/echocancel.h
index e0ceda0445a67dc636a89df8874b21367b6a0c54..f5b1b3d6824983eadc8b394ec7b1775d4e38d545 100644
--- a/sflphone-common/src/audio/echocancel.h
+++ b/sflphone-common/src/audio/echocancel.h
@@ -265,6 +265,17 @@ class EchoCancel : public Algorithm {
int _spkrHistoryLength;
+ /**
+ * Factor for power estimation
+ */
+ float _alpha;
+
+ /**
+ * Termporary spkr level memories
+ */
+ SFLDataFormat _spkrLevelMem;
+ SFLDataFormat _micLevelMem;
+
/**
* Current playback level
*/
@@ -285,13 +296,6 @@ class EchoCancel : public Algorithm {
*/
int _micHistCnt;
- /**
- * Average speaker/microphone level history. Each value corespond to
- * the averaged amplitude value over a segment (SEGMENT_LENGTH long)
- */
- int _avgSpkrLevelHist[BUFF_SIZE];
- int _avgMicLevelHist[BUFF_SIZE];
-
/**
* Current linear gain factor to be applied on microphone
*/
@@ -302,11 +306,6 @@ class EchoCancel : public Algorithm {
*/
float _lastAmplFactor;
- /**
- * Linear gain factor buffer to adjust to system's latency
- */
- float _delayLineAmplify[MAX_DELAY_LINE_AMPL];
-
/**
* read/write for mic gain delay
*/
@@ -328,17 +327,6 @@ class EchoCancel : public Algorithm {
*/
int _adaptCnt;
- /**
- * Factor for power estimation
- */
- float _alpha;
-
- /**
- * Termporary spkr level memories
- */
- SFLDataFormat _spkrLevelMem;
- SFLDataFormat _micLevelMem;
-
int _spkrAdaptCnt;
int _micAdaptCnt;
@@ -347,16 +335,38 @@ class EchoCancel : public Algorithm {
int _micAdaptSize;
+ int _correlationSize;
+
+ int _processedByte;
+
+ /**
+ * true if noise suppressor is active, false elsewhere
+ */
+ bool _echoActive;
+
+ /**
+ * true if noise suppressor is active, false elsewhere
+ */
+ bool _noiseActive;
+
+ /**
+ * Average speaker/microphone level history. Each value corespond to
+ * the averaged amplitude value over a segment (SEGMENT_LENGTH long)
+ */
+ int _avgSpkrLevelHist[BUFF_SIZE];
+ int _avgMicLevelHist[BUFF_SIZE];
+
+ /**
+ * Linear gain factor buffer to adjust to system's latency
+ */
+ float _delayLineAmplify[MAX_DELAY_LINE_AMPL];
+
int _spkrAdaptArray[BUFF_SIZE];
int _micAdaptArray[BUFF_SIZE];
- int _correlationSize;
-
int _correlationArray[BUFF_SIZE];
- int _processedByte;
-
ofstream *micFile;
ofstream *spkrFile;
ofstream *echoFile;
@@ -371,16 +381,6 @@ class EchoCancel : public Algorithm {
SpeexPreprocessState *_noiseState;
// #endif
- /**
- * true if noise suppressor is active, false elsewhere
- */
- bool _echoActive;
-
- /**
- * true if noise suppressor is active, false elsewhere
- */
- bool _noiseActive;
-
DelayDetection _delayDetector;
};
diff --git a/sflphone-common/src/audio/jitterbuf.cpp b/sflphone-common/src/audio/jitterbuf.cpp
index c72a6387f9769ae91feefd23e332030a74a0d328..510317cfeda83c538a6d5769eacfc173c6113960 100755
--- a/sflphone-common/src/audio/jitterbuf.cpp
+++ b/sflphone-common/src/audio/jitterbuf.cpp
@@ -123,7 +123,7 @@ static int longcmp(const void *a, const void *b)
\note maybe later we can make the history buckets variable size, or something? */
/* drop parameter determines whether we will drop outliers to minimize
* delay */
-static int history_put(jitterbuf *jb, long ts, long now, long ms)
+static int history_put(jitterbuf *jb, long ts, long now)
{
long delay = now - (ts - jb->info.resync_offset);
long threshold = 2 * jb->info.jitter + jb->info.conf.resync_threshold;
@@ -527,7 +527,7 @@ jb_return_code jb_put(jitterbuf *jb, void *data, const enum jb_frame_type type,
if (type == JB_TYPE_VOICE) {
/* presently, I'm only adding VOICE frames to history and drift calculations; mostly because with the
* IAX integrations, I'm sending retransmitted control frames with their awkward timestamps through */
- if (history_put(jb,ts,now,ms)) {
+ if (history_put(jb,ts,now)) {
jb->info.frames_dropped++;
return JB_DROP;
}
diff --git a/sflphone-common/src/audio/pulseaudio/audiostream.cpp b/sflphone-common/src/audio/pulseaudio/audiostream.cpp
index 41acf4c26362dcce9fa509f7d2e49ba257069009..8898fcdc39a8187dd46230c85d764b85c9a7bafa 100644
--- a/sflphone-common/src/audio/pulseaudio/audiostream.cpp
+++ b/sflphone-common/src/audio/pulseaudio/audiostream.cpp
@@ -68,7 +68,7 @@ AudioStream::connectStream(std::string* deviceName)
return true;
}
-static void success_cb (pa_stream *s, int success, void *userdata)
+static void success_cb (pa_stream *s, int success UNUSED, void *userdata)
{
assert (s);
@@ -78,7 +78,6 @@ static void success_cb (pa_stream *s, int success, void *userdata)
pa_threaded_mainloop_signal (mainloop, 0);
}
-
bool
AudioStream::drainStream (void)
{
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
index 667f0cf2728e4910298733b49692da4df4f6788f..0d9afc8c9d03f522de46c86733fc077765fbd923 100755
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -60,22 +60,12 @@ static void ringtone_callback (pa_stream* s, size_t bytes, void* userdata) {
}
-
-static void stream_moved_callback(pa_stream *s, void *userdata) {
-
- int streamIndex = pa_stream_get_index(s);
- int deviceIndex = pa_stream_get_device_index(s);
+static void stream_moved_callback(pa_stream *s, void *userdata UNUSED) {
_debug("stream_moved_callback: stream %d to %d", pa_stream_get_index(s), pa_stream_get_device_index(s));
-
-}
-
-static void pa_success_callback(pa_context *c, int success, void *userdata) {
-
- _debug("Audio: Success callback");
}
-static void latency_update_callback(pa_stream *p, void *userdata) {
+static void latency_update_callback(pa_stream *p, void *userdata UNUSED) {
pa_usec_t r_usec;
@@ -90,7 +80,7 @@ static void latency_update_callback(pa_stream *p, void *userdata) {
}
-static void sink_input_info_callback(pa_context *c, const pa_sink_info *i, int eol, void *userdata) {
+static void sink_input_info_callback(pa_context *c UNUSED, const pa_sink_info *i, int eol, void *userdata) {
char s[PA_SAMPLE_SPEC_SNPRINT_MAX], cv[PA_CVOLUME_SNPRINT_MAX], cm[PA_CHANNEL_MAP_SNPRINT_MAX];
if(!eol) {
@@ -124,10 +114,9 @@ static void sink_input_info_callback(pa_context *c, const pa_sink_info *i, int e
((PulseLayer *)userdata)->getSinkList()->push_back(deviceName);
}
-
}
-static void source_input_info_callback(pa_context *c, const pa_source_info *i, int eol, void *userdata) {
+static void source_input_info_callback(pa_context *c UNUSED, const pa_source_info *i, int eol, void *userdata) {
char s[PA_SAMPLE_SPEC_SNPRINT_MAX], cv[PA_CVOLUME_SNPRINT_MAX], cm[PA_CHANNEL_MAP_SNPRINT_MAX];
if(!eol) {
@@ -163,8 +152,7 @@ static void source_input_info_callback(pa_context *c, const pa_source_info *i, i
}
}
-
-static void context_changed_callback(pa_context* c, pa_subscription_event_type_t t, uint32_t idx, void* userdata)
+static void context_changed_callback(pa_context* c, pa_subscription_event_type_t t, uint32_t idx UNUSED, void* userdata UNUSED)
{
switch(t) {
@@ -220,28 +208,15 @@ static void context_changed_callback(pa_context* c, pa_subscription_event_type_t
_debug("Audio: Unknown event type");
}
-
}
-/*
-static void stream_suspended_callback (pa_stream *s UNUSED, void *userdata UNUSED)
+static void playback_underflow_callback (pa_stream* s UNUSED, void* userdata UNUSED)
{
- _debug("Audio: Stream Suspended");
-}
-*/
-
-
-static void playback_underflow_callback (pa_stream* s, void* userdata UNUSED)
-{
- // _debug ("Audio: Buffer Underflow");
- // pa_stream_trigger (s, NULL, NULL);
}
static void playback_overflow_callback (pa_stream* s UNUSED, void* userdata UNUSED)
{
- // _debug ("Audio: Buffer OverFlow");
-
}
@@ -654,15 +629,6 @@ int PulseLayer::canGetMic()
}
-int PulseLayer::getMic (void *buffer, int toCopy)
-{
- if (record) {
- return 0;
- } else
- return 0;
-}
-
-
void PulseLayer::startStream (void)
{
if(_audiofilter)
@@ -1012,8 +978,6 @@ void PulseLayer::readFromMic (void)
void PulseLayer::ringtoneToSpeaker(void)
{
- int availBytes;
-
AudioLoop* file_tone = _manager->getTelephoneFile();
SFLDataFormat* out;
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.h b/sflphone-common/src/audio/pulseaudio/pulselayer.h
index 047fe84093f133c5ee4a29ae9b676d993944d774..b2340dbe5e37d17f4d23d5d711ea99fad8cb0c48 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.h
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.h
@@ -100,14 +100,6 @@ class PulseLayer : public AudioLayer {
*/
int canGetMic();
- /**
- * Get data from the capture device
- * @param buffer The buffer for data
- * @param toCopy The number of bytes to get
- * @return int The number of bytes acquired ( 0 if an error occured)
- */
- int getMic(void *, int);
-
static void overflow ( pa_stream* s, void* userdata );
static void underflow ( pa_stream* s, void* userdata );
static void stream_state_callback( pa_stream* s, void* user_data );
diff --git a/sflphone-common/src/audio/sound/audiofile.cpp b/sflphone-common/src/audio/sound/audiofile.cpp
index 9bde16aea379a854e4fcc092b920a4bb01e6a3c9..04be6d1d6cd78b1984236f1c7286bd8cfd03acc2 100644
--- a/sflphone-common/src/audio/sound/audiofile.cpp
+++ b/sflphone-common/src/audio/sound/audiofile.cpp
@@ -174,157 +174,3 @@ AudioFile::loadFile (const std::string& filename, AudioCodec* codec , unsigned i
return true;
}
-
-
-
-WavFile::WavFile()
- : AudioLoop(),
- _filename(),
- _codec (NULL),
- _start (false)
-
-{
-}
-
-WavFile::~WavFile()
-{
-}
-
-
-bool WavFile::isFileExist(const std::string& filename)
-{
-
- std::fstream fs(filename.c_str(), std::ios_base::in);
- if(!fs) {
- _debug("WavFile: File \"%s\" doesn't exist", filename.c_str());
- return false;
- }
-
- return true;
-
-}
-
-bool WavFile::loadFile (const std::string& filename, AudioCodec* codec , unsigned int sampleRate)
-{
- if(!isFileExist(filename))
- return false;
-
- std::fstream file;
-
- printf("WavFile: Open %s", filename.c_str());
-
- file.open(filename.c_str(), std::ios::in | std::ios::binary);
-
- char riff[4] = {};
-
- file.read(riff, 4);
-
- if ( strncmp("RIFF", riff, 4) != 0 ) {
- _error("WavFile: File is not of RIFF format");
- return false;
- }
-
- // Find the "fmt " chunk
- char fmt[4] = {};
-
- while ( strncmp("fmt ", fmt, 4) != 0 ) {
- file.read(fmt, 4);
- _debug("WavFile: Searching... %s", fmt);
- }
-
- SINT32 chunkSize; // fmt chunk size
- unsigned short formatTag; // data compression tag
-
- file.read((char*)&chunkSize, 4); // Read fmt chunk size.
- file.read((char*)&formatTag, 2);
-
- _debug("Chunk size: %d\n", chunkSize);
- _debug("Format tag: %d\n", formatTag);
-
-
- if (formatTag != 1 ) { // PCM = 1, FLOAT = 3 {
- _error("WaveFile: File contains an unsupported data format type");
- return false;
- }
-
- // Get number of channels from the header.
- SINT16 chan;
- file.read((char*)&chan, 2);
-
- _channels = chan;
-
- _debug("WavFile: channel %d", _channels);
-
-
- // Get file sample rate from the header.
- SINT32 srate;
- file.read((char*)&srate, 4);
-
- _fileRate = (double)srate;
-
- printf("WavFile: srate %d", srate);
-
- SINT32 avgb;
- file.read((char*)&avgb, 4);
-
- _debug("WavFile: Average byte %i\n", avgb);
-
- SINT16 blockal;
- file.read((char*)&blockal, 2);
-
- _debug("WaveFile: block alignment %d", blockal);
-
- // Determine the data type
- _dataType = 0;
-
- SINT16 dt;
- file.read((char*)&dt, 2);
-
- _debug("WaveFile: dt %d", dt);
-
- if ( formatTag == 1 ) {
- if (dt == 8)
- _dataType = 1; // SINT8;
- else if (dt == 16)
- _dataType = 2; // SINT16;
- else if (dt == 32)
- _dataType = 3; // SINT32;
- }
-
- /*
- else if ( formatTag == 3 )
- {
- if (temp == 32)
- dataType_ = FLOAT32;
- else if (temp == 64)
- dataType_ = FLOAT64;
- }
- */
- else {
- _debug("WavFile: File's bits per sample is not supported");
- return false;
- }
-
- // Find the "data" chunk
- char data[4] = {};
-
- while ( strncmp("data", data, 4) ) {
- file.read(data, 4);
- _debug("WavFile: Searching data");
- }
-
- // Get length of data from the header.
- SINT32 bytes;
- file.read((char*)&bytes, 4);
-
- _debug("WavFile: Data size in byte %ld", bytes);
-
- _fileSize = 8 * bytes / dt / _channels; // sample frames
-
- _debug("WavFile: Data size in frames %ld", _fileSize);
-
- _debug("WavFile: File successfully opened");
-
- return true;
-
-}
diff --git a/sflphone-common/src/audio/sound/audiofile.h b/sflphone-common/src/audio/sound/audiofile.h
index cb96561ff2b6f3a748d830442ffa72de9cedfc64..019783defe16be7a76ecc16f97f02c3bc1eeec69 100644
--- a/sflphone-common/src/audio/sound/audiofile.h
+++ b/sflphone-common/src/audio/sound/audiofile.h
@@ -100,76 +100,5 @@ private:
bool _start;
};
+#endif
-
-
-/**
- * @file audiofile.h
- * @brief A class to manage wave files
- */
-
-class WavFile : public AudioLoop
-{
-public:
- /**
- * Constructor
- */
- WavFile();
-
- /**
- * Destructor
- */
- ~WavFile();
-
-
- /**
- * Load a sound file in memory
- * @param filename The absolute path to the file
- * @param codec The codec to decode and encode it
- * @param sampleRate The sample rate to read it
- * @return bool True on success
- */
- bool loadFile(const std::string& filename, AudioCodec *codec , unsigned int sampleRate = 44100);
-
- /**
- * Start the sound file
- */
- void start() { _start = true; }
-
- /**
- * Stop the sound file
- */
- void stop() { _start = false; }
-
- /**
- * Tells whether or not the file is playing
- * @return bool True if yes
- * false otherwise
- */
- bool isStarted() { return _start; }
-
-private:
- // Copy Constructor
- WavFile(const AudioFile& rh);
-
- // Assignment Operator
- WavFile& operator=( const AudioFile& rh);
-
- bool isFileExist(const std::string& filename);
-
- /** The absolute path to the sound file */
- std::string _filename;
-
- /** Your preferred codec */
- AudioCodec* _codec;
-
- /** Start or not */
- bool _start;
-
- int _channels;
- int _fileRate;
- int _dataType;
- int _fileSize;
-};
-
-#endif // __AUDIOFILE_H__
diff --git a/sflphone-common/src/audio/speexechocancel.cpp b/sflphone-common/src/audio/speexechocancel.cpp
index fa5f1045ddfd0566d8ed36e018a9fcbf58f9df28..c956af0f9774cb253e8fcbc8954612cc3cfa72cc 100644
--- a/sflphone-common/src/audio/speexechocancel.cpp
+++ b/sflphone-common/src/audio/speexechocancel.cpp
@@ -100,7 +100,7 @@ void SpeexEchoCancel::putData(SFLDataFormat *inputData, int nbBytes)
// speex_echo_playback(_echoState, inputData);
}
-void SpeexEchoCancel::process(SFLDataFormat *data, int nbBytes) {}
+void SpeexEchoCancel::process(SFLDataFormat *data UNUSED, int nbBytes UNUSED) {}
int SpeexEchoCancel::process(SFLDataFormat *inputData, SFLDataFormat *outputData, int nbBytes)
{
@@ -158,9 +158,6 @@ int SpeexEchoCancel::process(SFLDataFormat *inputData, SFLDataFormat *outputData
return nbFrame * FRAME_SIZE;
}
-void SpeexEchoCancel::process(SFLDataFormat *micData, SFLDataFormat *spkrData, SFLDataFormat *outputData, int nbBytes){
-
-
- // speex_echo_cancellation(_echoState, micData, spkrData, outputData);
+void SpeexEchoCancel::process(SFLDataFormat *micData UNUSED, SFLDataFormat *spkrData UNUSED, SFLDataFormat *outputData UNUSED, int nbBytes UNUSED){
}
diff --git a/sflphone-common/src/config/yamlengine.cpp b/sflphone-common/src/config/yamlengine.cpp
index 30a2545bfdea02643916d02cbbef5565196335fa..10022a189dcf817c5e6547b5eb9d519c5807a59c 100755
--- a/sflphone-common/src/config/yamlengine.cpp
+++ b/sflphone-common/src/config/yamlengine.cpp
@@ -41,7 +41,7 @@ YamlEngine::~YamlEngine() {}
void YamlEngine::openConfigFile()
{
- Conf::YamlParser *parser;
+ Conf::YamlParser *parser = NULL;
try {
parser = new Conf::YamlParser("sequence.yml");
diff --git a/sflphone-common/src/iax/iaxvoiplink.cpp b/sflphone-common/src/iax/iaxvoiplink.cpp
index 415bda1f6eb01bdeafeb5a5a6a925ca2c51b1baa..6e2f09eaea1283f3112e50b31f7d5614d876a3a7 100644
--- a/sflphone-common/src/iax/iaxvoiplink.cpp
+++ b/sflphone-common/src/iax/iaxvoiplink.cpp
@@ -408,7 +408,7 @@ IAXVoIPLink::getIAXCall (const CallID& id)
int
-IAXVoIPLink::sendRegister (AccountID id)
+IAXVoIPLink::sendRegister (AccountID id UNUSED)
{
IAXAccount *account;
bool result;
@@ -459,7 +459,7 @@ IAXVoIPLink::sendRegister (AccountID id)
}
int
-IAXVoIPLink::sendUnregister (AccountID id)
+IAXVoIPLink::sendUnregister (AccountID id UNUSED)
{
IAXAccount *account;
diff --git a/sflphone-common/src/managerimpl.cpp b/sflphone-common/src/managerimpl.cpp
index 8652d731c291214d45b0ba62d9ee5b837b22b2e4..f946e4ed561d7c0123362e1b23ac00a38e6f5509 100755
--- a/sflphone-common/src/managerimpl.cpp
+++ b/sflphone-common/src/managerimpl.cpp
@@ -3224,72 +3224,10 @@ std::string ManagerImpl::computeMd5HashFromCredential (
return hashedDigest;
}
-void ManagerImpl::setCredential (const std::string& accountID,
- const int32_t& index, const std::map<std::string, std::string>& details) {
+void ManagerImpl::setCredential (const std::string& accountID UNUSED,
+ const int32_t& index UNUSED, const std::map<std::string UNUSED, std::string>& details UNUSED) {
_debug("Manager: set credential");
- /*
- std::map<std::string, std::string>::iterator it;
- std::map<std::string, std::string> credentialInformation = details;
-
- std::string credentialIndex;
- std::stringstream streamOut;
- streamOut << index;
- credentialIndex = streamOut.str();
-
- std::string section = "Credential" + std::string(":") + accountID
- + std::string(":") + credentialIndex;
-
- _debug ("Setting credential in section %s", section.c_str());
-
- it = credentialInformation.find(USERNAME);
- std::string username;
-
- if (it == credentialInformation.end()) {
- username = EMPTY_FIELD;
- } else {
- username = it->second;
- }
-
- Manager::instance().setConfig(section, USERNAME, username);
-
- it = credentialInformation.find(REALM);
- std::string realm;
-
- if (it == credentialInformation.end()) {
- realm = EMPTY_FIELD;
- } else {
- realm = it->second;
- }
-
- Manager::instance().setConfig(section, REALM, realm);
-
- it = credentialInformation.find(PASSWORD);
- std::string password;
-
- if (it == credentialInformation.end()) {
- password = EMPTY_FIELD;
- } else {
- password = it->second;
- }
-
- if (getMd5CredentialHashing()) {
- // TODO: Fix this.
- // This is an extremly weak test in order to check
- // if the password is a hashed value. This is done
- // because deleteCredential() is called before this
- // method. Therefore, we cannot check if the value
- // is different from the one previously stored in
- // the configuration file. This is to avoid to
- // re-hash a hashed password.
-
- if (password.length() != 32) {
- password = computeMd5HashFromCredential(username, password, realm);
- }
- }
-
- Manager::instance().setConfig(section, PASSWORD, password);
- */
}
// method to reduce the if/else mess.
diff --git a/sflphone-common/src/sip/sipaccount.cpp b/sflphone-common/src/sip/sipaccount.cpp
index 4a9bfb8b88a53b67b73de87cbd4024ba7e133263..6c30a0d5fd3abece237d4bb8d8932fa1698a0ff6 100755
--- a/sflphone-common/src/sip/sipaccount.cpp
+++ b/sflphone-common/src/sip/sipaccount.cpp
@@ -36,10 +36,6 @@
#include <pwd.h>
#include <sstream>
-// CredentialItem::CredentialItem() {}
-
-// CredentialItem::~CredentialItem() {}
-
Credentials::Credentials() : credentialCount(0) {}
Credentials::~Credentials() {}
@@ -60,7 +56,7 @@ CredentialItem *Credentials::getCredential(int index)
return NULL;
}
-void Credentials::serialize(Conf::YamlEmitter *emitter)
+void Credentials::serialize(Conf::YamlEmitter *emitter UNUSED)
{
}
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index 45f8a1437efec7765417139b049765498c9ef301..c5df6f3465f60e779ff892ff66c77ab1fff0d520 100755
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -1594,7 +1594,7 @@ SIPVoIPLink::SIPCallReleased (SIPCall *call)
void
-SIPVoIPLink::SIPCallAnswered (SIPCall *call, pjsip_rx_data *rdata)
+SIPVoIPLink::SIPCallAnswered (SIPCall *call, pjsip_rx_data *rdata UNUSED)
{
_info ("UserAgent: SIP call answered");
@@ -3021,27 +3021,9 @@ void set_voicemail_info (AccountID account, pjsip_msg_body *body)
Manager::instance().startVoiceMessageNotification (account, voicemail);
}
-void SIPVoIPLink::handle_reinvite (SIPCall *call)
+void SIPVoIPLink::handle_reinvite (SIPCall *call UNUSED)
{
-
_debug ("UserAgent: Handle reinvite");
- /*
- // Close the previous RTP session
- call->getAudioRtp()->stop ();
- call->setAudioStart (false);
-
- _debug ("Create new rtp session from handle_reinvite : %s:%i", call->getLocalIp().c_str(), call->getLocalAudioPort());
- _debug ("UserAgent: handle_reinvite");
-
- try {
- call->getAudioRtp()->initAudioRtpSession (call);
- } catch (...) {
- _debug ("! SIP Failure: Unable to create RTP Session (%s:%d)", __FILE__, __LINE__);
- }
-
-
- _debug("Handle reINVITE");
- */
}
// This callback is called when the invite session state has changed
@@ -3059,8 +3041,6 @@ void call_on_state_changed (pjsip_inv_session *inv, pjsip_event *e)
_error ("UserAgent: Error: Call is NULL in call state changed callback");
return;
} else {
- // _debug(" call_on_state_changed: call id %s", call->getCallId().c_str());
- // _debug(" call_on_state_changed: call state %s", invitationStateMap[call->getInvSession()->state]);
}
//Retrieve the body message
@@ -3364,11 +3344,11 @@ void call_on_media_update (pjsip_inv_session *inv, pj_status_t status)
}
-void call_on_forked (pjsip_inv_session *inv, pjsip_event *e)
+void call_on_forked (pjsip_inv_session *inv UNUSED, pjsip_event *e UNUSED)
{
}
-void call_on_tsx_changed (pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e)
+void call_on_tsx_changed (pjsip_inv_session *inv UNUSED, pjsip_transaction *tsx, pjsip_event *e)
{
assert(tsx);
@@ -4263,7 +4243,7 @@ void xfer_svr_cb (pjsip_evsub *sub, pjsip_event *event)
}
}
-void on_rx_offer (pjsip_inv_session *inv, const pjmedia_sdp_session *offer)
+void on_rx_offer (pjsip_inv_session *inv, const pjmedia_sdp_session *offer UNUSED)
{
_info("UserAgent: Received SDP offer");
@@ -4540,7 +4520,7 @@ std::vector<std::string> SIPVoIPLink::getAllIpInterfaceByName(void)
}
-pj_bool_t stun_sock_on_status (pj_stun_sock *stun_sock, pj_stun_sock_op op, pj_status_t status)
+pj_bool_t stun_sock_on_status (pj_stun_sock *stun_sock UNUSED, pj_stun_sock_op op UNUSED, pj_status_t status)
{
if (status == PJ_SUCCESS)
return PJ_TRUE;
@@ -4548,7 +4528,7 @@ pj_bool_t stun_sock_on_status (pj_stun_sock *stun_sock, pj_stun_sock_op op, pj_s
return PJ_FALSE;
}
-pj_bool_t stun_sock_on_rx_data (pj_stun_sock *stun_sock, void *pkt, unsigned pkt_len, const pj_sockaddr_t *src_addr, unsigned addr_len)
+pj_bool_t stun_sock_on_rx_data (pj_stun_sock *stun_sock UNUSED, void *pkt UNUSED, unsigned pkt_len UNUSED, const pj_sockaddr_t *src_addr UNUSED, unsigned addr_len UNUSED)
{
return PJ_TRUE;
}