diff --git a/daemon/src/audio/audiorecord.cpp b/daemon/src/audio/audiorecord.cpp
index 3da2522a8..c59e15407 100644
--- a/daemon/src/audio/audiorecord.cpp
+++ b/daemon/src/audio/audiorecord.cpp
@@ -420,7 +420,7 @@ void AudioRecord::closeWavFile()
         WARN("Can't close file");
 }
 
-//void AudioRecord::recData(SFLDataFormat* buffer, size_t nSamples)
+//void AudioRecord::recData(SFLAudioSample* buffer, size_t nSamples)
 void AudioRecord::recData(AudioBuffer& buffer)
 {
     if (recordingEnabled_) {
diff --git a/daemon/src/audio/codecs/audiocodec.h b/daemon/src/audio/codecs/audiocodec.h
index 5f7e29b11..c1d4e0269 100644
--- a/daemon/src/audio/codecs/audiocodec.h
+++ b/daemon/src/audio/codecs/audiocodec.h
@@ -36,6 +36,7 @@
 #include <vector>
 
 #include "cc_config.h"
+#include "sfl_types.h"
 #include <ccrtp/formats.h> // for ost::DynamicPayloadFormat
 
 #define XSTR(s) STR(s)
@@ -70,14 +71,14 @@ class AudioCodec {
          * @param buffer_size : the size of the input buffer
          * @return the number of samples decoded
          */
-        virtual int decode(short *dst, unsigned char *buf, size_t buffer_size) = 0;
+        virtual int decode(SFLAudioSample *dst, unsigned char *buf, size_t buffer_size) = 0;
 
         /**
          * Encode an input buffer and fill the output buffer with the encoded data
          * @param buffer_size : the maximum size of encoded data buffer (dst)
          * @return the number of bytes encoded
          */
-        virtual int encode(unsigned char *dst, short *src, size_t buffer_size) = 0;
+        virtual int encode(unsigned char *dst, SFLAudioSample *src, size_t buffer_size) = 0;
 
         /**
          * Multichannel version of decode().
diff --git a/daemon/src/audio/codecs/g729.cpp b/daemon/src/audio/codecs/g729.cpp
index 8d82c8795..217b21d6f 100644
--- a/daemon/src/audio/codecs/g729.cpp
+++ b/daemon/src/audio/codecs/g729.cpp
@@ -28,12 +28,13 @@
  *  as that of the covered work.
  */
 #include "g729.h"
+#include "sfl_types.h"
 #include <iostream>
 #include <dlfcn.h>
 #include <stdexcept>
 
-#define G729_TYPE_ENCODER        (void (*)(bcg729EncoderChannelContextStruct*, int16_t[], uint8_t[]))
-#define G729_TYPE_DECODER        (void (*)(bcg729DecoderChannelContextStruct*, uint8_t[], uint8_t, int16_t[]))
+#define G729_TYPE_ENCODER        (void (*)(bcg729EncoderChannelContextStruct*, SFLAudioSample[], uint8_t[]))
+#define G729_TYPE_DECODER        (void (*)(bcg729DecoderChannelContextStruct*, uint8_t[], uint8_t, SFLAudioSample[]))
 
 #define G729_TYPE_DECODER_INIT   (bcg729DecoderChannelContextStruct*(*)())
 #define G729_TYPE_ENCODER_INIT   (bcg729EncoderChannelContextStruct*(*)())
@@ -72,14 +73,14 @@ G729::~G729()
         dlclose(handler_);
 }
 
-int G729::decode(short *dst, unsigned char *buf, size_t buffer_size)
+int G729::decode(SFLAudioSample *dst, unsigned char *buf, size_t buffer_size)
 {
     decoder_(decoderContext_, buf, false, dst);
     decoder_(decoderContext_, buf + (buffer_size / 2), false, dst + 80);
     return 160;
 }
 
-int G729::encode(unsigned char *dst, short *src, size_t buffer_size)
+int G729::encode(unsigned char *dst, SFLAudioSample *src, size_t buffer_size)
 {
     encoder_(encoderContext_, src, dst);
     encoder_(encoderContext_, src + (buffer_size / 2), dst + 10);
diff --git a/daemon/src/audio/codecs/g729.h b/daemon/src/audio/codecs/g729.h
index 11d83c0d8..1c2671b74 100644
--- a/daemon/src/audio/codecs/g729.h
+++ b/daemon/src/audio/codecs/g729.h
@@ -31,6 +31,7 @@
 #define G729_H_
 
 #include <cstdlib>
+#include "sfl_types.h"
 #include "noncopyable.h"
 
 #include "audiocodec.h"
@@ -43,8 +44,8 @@ public:
    G729();
    ~G729();
 private:
-   virtual int decode(short *dst, unsigned char *buf, size_t buffer_size);
-   virtual int encode(unsigned char *dst, short *src, size_t buffer_size);
+   virtual int decode(SFLAudioSample *dst, unsigned char *buf, size_t buffer_size);
+   virtual int encode(unsigned char *dst, SFLAudioSample *src, size_t buffer_size);
 
    NON_COPYABLE(G729);
    //Attributes
@@ -53,8 +54,8 @@ private:
    void* handler_;
 
    //Extern functions
-   void (*encoder_) (bcg729EncoderChannelContextStruct *encoderChannelContext, int16_t inputFrame[], uint8_t bitStream[]);
-   void (*decoder_) (bcg729DecoderChannelContextStruct *decoderChannelContext, uint8_t bitStream[], uint8_t frameErasureFlag, int16_t signal[]);
+   void (*encoder_) (bcg729EncoderChannelContextStruct *encoderChannelContext, SFLAudioSample inputFrame[], uint8_t bitStream[]);
+   void (*decoder_) (bcg729DecoderChannelContextStruct *decoderChannelContext, uint8_t bitStream[], uint8_t frameErasureFlag, SFLAudioSample signal[]);
 
    static void loadError(const char *error);
 };
diff --git a/daemon/src/audio/codecs/ilbc.cpp b/daemon/src/audio/codecs/ilbc.cpp
index d7ccf0932..1fb5a2835 100644
--- a/daemon/src/audio/codecs/ilbc.cpp
+++ b/daemon/src/audio/codecs/ilbc.cpp
@@ -29,6 +29,7 @@
  */
 
 #include "audiocodec.h"
+#include "sfl_types.h"
 #include <tr1/array>
 #include <algorithm>
 
@@ -50,13 +51,13 @@ class Ilbc: public sfl::AudioCodec {
 
     private:
         // iLBC expects floating point data, so we have to convert
-        int decode(short *dst, unsigned char *src, size_t /*buf_size*/) {
+        int decode(SFLAudioSample *dst, unsigned char *src, size_t /*buf_size*/) {
             const int NORMAL_MODE = 1;
             iLBC_decode(dst, reinterpret_cast<WebRtc_UWord16*>(src), &ilbc_dec_, NORMAL_MODE);
             return frameSize_;
         }
 
-        int encode(unsigned char *dst, short* src, size_t /*buf_size*/) {
+        int encode(unsigned char *dst, SFLAudioSample * src, size_t /*buf_size*/) {
             iLBC_encode(reinterpret_cast<WebRtc_UWord16*>(dst), src, &ilbc_enc_);
             return frameSize_;
         }
diff --git a/daemon/src/audio/codecs/opus.cpp b/daemon/src/audio/codecs/opus.cpp
index 74ac21e57..f447c8e2a 100644
--- a/daemon/src/audio/codecs/opus.cpp
+++ b/daemon/src/audio/codecs/opus.cpp
@@ -29,6 +29,7 @@
  *  as that of the covered work.
  */
 #include "opus.h"
+#include "sfl_types.h"
 #include <stdexcept>
 #include <iostream>
 
@@ -60,12 +61,12 @@ Opus::~Opus()
         opus_decoder_destroy(decoder_);
 }
 
-int Opus::decode(short *dst, unsigned char *buf, size_t buffer_size)
+int Opus::decode(SFLAudioSample *dst, unsigned char *buf, size_t buffer_size)
 {
     return opus_decode(decoder_, buf, buffer_size, dst, FRAME_SIZE, 0);
 }
 
-int Opus::encode(unsigned char *dst, short *src, size_t buffer_size)
+int Opus::encode(unsigned char *dst, SFLAudioSample *src, size_t buffer_size)
 {
     return opus_encode(encoder_, src, FRAME_SIZE, dst, buffer_size * 2);
 }
diff --git a/daemon/src/audio/codecs/opus.h b/daemon/src/audio/codecs/opus.h
index f4064fbd0..fe3d30f48 100644
--- a/daemon/src/audio/codecs/opus.h
+++ b/daemon/src/audio/codecs/opus.h
@@ -32,6 +32,7 @@
 #define OPUS_H_
 
 #include "noncopyable.h"
+#include "sfl_types.h"
 
 #include "audiocodec.h"
 
@@ -45,8 +46,8 @@ public:
    static const uint8 PAYLOAD_TYPE = 104; // dynamic payload type, out of range of video (96-99)
 
 private:
-   virtual int decode(short *dst, unsigned char *buf, size_t buffer_size);
-   virtual int encode(unsigned char *dst, short *src, size_t buffer_size);
+   virtual int decode(SFLAudioSample *dst, unsigned char *buf, size_t buffer_size);
+   virtual int encode(unsigned char *dst, SFLAudioSample *src, size_t buffer_size);
 
    //multichannel version
    virtual int decode(std::vector<std::vector<short> > *dst, unsigned char *buf, size_t buffer_size, size_t dst_offset=0);
diff --git a/daemon/src/audio/codecs/speexcodec.h b/daemon/src/audio/codecs/speexcodec.h
index 20b1b66a0..6494d57cd 100644
--- a/daemon/src/audio/codecs/speexcodec.h
+++ b/daemon/src/audio/codecs/speexcodec.h
@@ -29,7 +29,7 @@
  *  as that of the covered work.
  */
 
-#include "global.h"
+#include "sfl_types.h"
 #include "audiocodec.h"
 #include "noncopyable.h"
 #include "array_size.h"
@@ -83,13 +83,13 @@ private:
 
         NON_COPYABLE(Speex);
 
-        virtual int decode(short *dst, unsigned char *src, size_t buf_size) {
+        virtual int decode(SFLAudioSample *dst, unsigned char *src, size_t buf_size) {
             speex_bits_read_from(&speex_dec_bits_, (char*) src, buf_size);
             speex_decode_int(speex_dec_state_, &speex_dec_bits_, dst);
             return frameSize_;
         }
 
-        virtual int encode(unsigned char *dst, short *src, size_t buf_size) {
+        virtual int encode(unsigned char *dst, SFLAudioSample *src, size_t buf_size) {
             speex_bits_reset(&speex_enc_bits_);
             speex_encode_int(speex_enc_state_, src, &speex_enc_bits_);
             return speex_bits_write(&speex_enc_bits_, (char*) dst, buf_size);
diff --git a/daemon/src/audio/dcblocker.cpp b/daemon/src/audio/dcblocker.cpp
index ab8c4037f..92e631ba5 100644
--- a/daemon/src/audio/dcblocker.cpp
+++ b/daemon/src/audio/dcblocker.cpp
@@ -39,7 +39,7 @@ void DcBlocker::reset()
     states.assign(states.size(), (struct StreamState){0, 0, 0, 0});
 }
 
-void DcBlocker::doProcess(SFLAudioSample *out, SFLAudioSample *in, int samples, struct StreamState * state)
+void DcBlocker::doProcess(SFLAudioSample *out, SFLAudioSample *in, unsigned samples, struct StreamState * state)
 {
     for (unsigned i = 0; i < samples; ++i) {
         state->x_ = in[i];
diff --git a/daemon/src/audio/dcblocker.h b/daemon/src/audio/dcblocker.h
index 4326222c8..878b6855a 100644
--- a/daemon/src/audio/dcblocker.h
+++ b/daemon/src/audio/dcblocker.h
@@ -52,7 +52,7 @@ class DcBlocker {
             SFLAudioSample y_, x_, xm1_, ym1_;
         };
 
-        void doProcess(SFLAudioSample *out, SFLAudioSample *in, int samples, struct StreamState * state);
+        void doProcess(SFLAudioSample *out, SFLAudioSample *in, unsigned samples, struct StreamState * state);
 
         std::vector<StreamState> states;
 };
diff --git a/daemon/src/audio/gaincontrol.cpp b/daemon/src/audio/gaincontrol.cpp
index 5e5beeea9..be7a98344 100644
--- a/daemon/src/audio/gaincontrol.cpp
+++ b/daemon/src/audio/gaincontrol.cpp
@@ -69,7 +69,7 @@ void GainControl::process(SFLAudioSample *buf, int samples)
 
     for (int i = 0; i < samples; i++) {
         // linear conversion
-        in = (double)buf[i] / (double)SHRT_MAX;
+        in = (double) buf[i] / (double) SFL_DATA_FORMAT_MAX;
 
         out = currentGain_ * in;
 
@@ -81,7 +81,7 @@ void GainControl::process(SFLAudioSample *buf, int samples)
 
         out = limiter_.limit(out);
 
-        buf[i] = (short)(out * (double)SHRT_MAX);
+        buf[i] = (SFLAudioSample) (out * (double) SFL_DATA_FORMAT_MAX);
     }
 
     diffRms = maxRms - targetLevelLinear_;
diff --git a/daemon/src/audio/mainbuffer.cpp b/daemon/src/audio/mainbuffer.cpp
index 19ec88cca..fb16a96ca 100644
--- a/daemon/src/audio/mainbuffer.cpp
+++ b/daemon/src/audio/mainbuffer.cpp
@@ -275,14 +275,14 @@ size_t MainBuffer::getData(AudioBuffer& buffer, const std::string &call_id)
 
         for (CallIDSet::iterator iter_id = callid_set->begin();
              iter_id != callid_set->end(); ++iter_id) {
-            /*size_t nbSmplToCopy = toCopy / sizeof(SFLDataFormat);
-            SFLDataFormat mixBuffer[nbSmplToCopy];
+            /*size_t nbSmplToCopy = toCopy / sizeof(SFLAudioSample);
+            SFLAudioSample mixBuffer[nbSmplToCopy];
             memset(mixBuffer, 0, toCopy);
             size = getDataByID(mixBuffer, toCopy, *iter_id, call_id);*/
             size = getDataByID(mixBuffer, *iter_id, call_id);
 
             if (size > 0) {
-                /*SFLDataFormat *dest = static_cast<SFLDataFormat*>(buffer);
+                /*SFLAudioSample *dest = static_cast<SFLAudioSample*>(buffer);
                 for (size_t k = 0; k < nbSmplToCopy; ++k)
                     dest[k] += mixBuffer[k];*/
                 buffer.mix(mixBuffer);
diff --git a/daemon/src/audio/samplerateconverter.cpp b/daemon/src/audio/samplerateconverter.cpp
index 0924ee8f5..83533d79f 100644
--- a/daemon/src/audio/samplerateconverter.cpp
+++ b/daemon/src/audio/samplerateconverter.cpp
@@ -29,6 +29,7 @@
  */
 
 #include "samplerateconverter.h"
+#include "sfl_types.h"
 #include "manager.h"
 #include <cassert>
 #include "logger.h"
@@ -54,7 +55,7 @@ SamplerateConverter::~SamplerateConverter()
 }
 
 void
-SamplerateConverter::Short2FloatArray(const short *in, float *out, int len)
+SamplerateConverter::Short2FloatArray(const SFLAudioSample *in, float *out, int len)
 {
     // factor is 1/(2^15), used to rescale the short int range to the
     // [-1.0 - 1.0] float range.
@@ -64,8 +65,8 @@ SamplerateConverter::Short2FloatArray(const short *in, float *out, int len)
 }
 
 void SamplerateConverter::resample(const AudioBuffer &dataIn, AudioBuffer &dataOut)
-/*void SamplerateConverter::resample(SFLDataFormat *dataIn,
-                                   SFLDataFormat *dataOut,
+/*void SamplerateConverter::resample(SFLAudioSample *dataIn,
+                                   SFLAudioSample *dataOut,
                                    size_t dataOutSize,
                                    int inputFreq,
                                    int outputFreq,
diff --git a/daemon/src/audio/samplerateconverter.h b/daemon/src/audio/samplerateconverter.h
index c379beba1..12099bc99 100644
--- a/daemon/src/audio/samplerateconverter.h
+++ b/daemon/src/audio/samplerateconverter.h
@@ -60,7 +60,7 @@ class SamplerateConverter {
          * @param SamplerateConverter2 The desired sample rate
          * @param nbSamples	  The number of samples to process
          */
-        //void resample(SFLDataFormat* dataIn, SFLDataFormat* dataOut, size_t dataOutSize, int oldrate, int newrate, size_t nbSamples);
+        //void resample(SFLAudioSample* dataIn, SFLAudioSample* dataOut, size_t dataOutSize, int oldrate, int newrate, size_t nbSamples);
         void resample(const AudioBuffer& dataIn, AudioBuffer& dataOut);
 
         /**
@@ -69,7 +69,7 @@ class SamplerateConverter {
          * @param out The resulting (float) array
          * @param len The number of elements in both tables
          */
-        void Short2FloatArray(const short *in, float *out, int len);
+        void Short2FloatArray(const SFLAudioSample *in, float *out, int len);
 
 
     private:
diff --git a/daemon/src/audio/sound/dtmfgenerator.h b/daemon/src/audio/sound/dtmfgenerator.h
index c5b9edb34..a9f600d79 100644
--- a/daemon/src/audio/sound/dtmfgenerator.h
+++ b/daemon/src/audio/sound/dtmfgenerator.h
@@ -101,7 +101,7 @@ class DTMFGenerator {
 
         /*
          * Get n samples of the signal of code code
-         * @param buffer a SFLDataFormat vector
+         * @param buffer a SFLAudioSample vector
          * @param code   dtmf code to get sound
          */
         void getSamples(std::vector<SFLAudioSample> &buffer, unsigned char code);
@@ -109,7 +109,7 @@ class DTMFGenerator {
         /*
          * Get next n samples (continues where previous call to
          * genSample or genNextSamples stopped
-         * @param buffer a SFLDataFormat vector
+         * @param buffer a SFLAudioSample vector
          */
         void getNextSamples(std::vector<SFLAudioSample> &buffer);
 
@@ -118,7 +118,7 @@ class DTMFGenerator {
         /**
          * Fill tone buffer for a given index of the array of tones.
          * @param index of the tone in the array tones_
-         * @return SFLDataFormat* The generated data
+         * @return SFLAudioSample* The generated data
          */
         SFLAudioSample* fillToneBuffer(int index);
 };
diff --git a/daemon/src/sfl_types.h b/daemon/src/sfl_types.h
index 50abb6e7c..a8993a619 100644
--- a/daemon/src/sfl_types.h
+++ b/daemon/src/sfl_types.h
@@ -32,12 +32,13 @@
 #define SFL_TYPES_H_
 
 #include <cstddef> // for size_t
+#include <stdint.h>
 
-typedef short SFLAudioSample;
-//typedef size_t AudioChannel;
+typedef int16_t SFLAudioSample;
 
-typedef signed short SINT16;
-typedef signed int SINT32;
+typedef int16_t SINT16;
+typedef int32_t SINT32;
+#define SFL_DATA_FORMAT_MAX SHRT_MAX
 
 /*
 typedef struct {