diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
index 8a5503ecf1f781cf765286a18a6059ea077f741b..5e53f507033fb4d07b09af555ad576e42ac74c47 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
@@ -159,7 +159,6 @@ int AudioRtpRecordHandler::processDataEncode (void)
int bytesToGet = samplesToGet * sizeof (SFLDataFormat);
if (Manager::instance().getMainBuffer()->availForGet (id_) < bytesToGet) {
- _error("%s : not enough data available", __PRETTY_FUNCTION__);
return 0;
}
@@ -190,14 +189,17 @@ int AudioRtpRecordHandler::processDataEncode (void)
}
_audioRtpRecord.audioCodecMutex.enter();
- int compSize = _audioRtpRecord._audioCodec->encode (micDataEncoded, out, getCodecFrameSize());
+ int compSize = _audioRtpRecord._audioCodec->encode (micDataEncoded, out, DEC_BUFFER_SIZE);
_audioRtpRecord.audioCodecMutex.leave();
return compSize;
}
-void AudioRtpRecordHandler::processDataDecode (unsigned char *spkrData, unsigned int size)
+void AudioRtpRecordHandler::processDataDecode (unsigned char *spkrData, unsigned int size, int payloadType)
{
+ if (getCodecPayloadType() != payloadType)
+ return;
+
int codecSampleRate = getCodecSampleRate();
SFLDataFormat *spkrDataDecoded = _audioRtpRecord.decData;
@@ -207,13 +209,11 @@ void AudioRtpRecordHandler::processDataDecode (unsigned char *spkrData, unsigned
_audioRtpRecord.audioCodecMutex.enter();
- // Return the size of data in bytes
- int expandedSize = _audioRtpRecord._audioCodec->decode (spkrDataDecoded , spkrData , size);
+ // Return the size of data in samples
+ int inSamples = _audioRtpRecord._audioCodec->decode (spkrDataDecoded , spkrData , size);
_audioRtpRecord.audioCodecMutex.leave();
- int inSamples = expandedSize / sizeof (SFLDataFormat);
-
fadeIn (spkrDataDecoded, inSamples, &_audioRtpRecord._micAmplFactor);
// Normalize incomming signal
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h
index c9d64998660d61db534146a7dd44c854a8c90453..eb35e07cb81d1f5bb6928560e7ec7c14f4f3305b 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h
@@ -144,7 +144,7 @@ class AudioRtpRecordHandler
return _audioRtpRecord._codecFrameSize;
}
- int getHasDynamicPayload (void) const {
+ bool getHasDynamicPayload (void) const {
return _audioRtpRecord._hasDynamicPayloadType;
}
@@ -172,7 +172,7 @@ class AudioRtpRecordHandler
/**
* Decode audio data received from peer
*/
- void processDataDecode (unsigned char * spkrData, unsigned int size);
+ void processDataDecode (unsigned char * spkrData, unsigned int size, int payloadType);
/**
* Ramp In audio data to avoid audio click from peer
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
index 4c75dc3ae77b48dfad87746c03154ef2c0a1a9fa..db1c2d1b1f91d41cd0abb275bbb2ec13cf050fc3 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
@@ -69,12 +69,13 @@ void AudioRtpSession::updateSessionMedia (AudioCodec *audioCodec)
// Update internal codec for this session
updateRtpMedia (audioCodec);
+ // store codec info locally
int payloadType = getCodecPayloadType();
int frameSize = getCodecFrameSize();
int smplRate = getCodecSampleRate();
- int dynamic = getHasDynamicPayload();
+ bool dynamic = getHasDynamicPayload();
- // G722 requires timetamp to be incremented at 8khz
+ // G722 requires timestamp to be incremented at 8khz
if (payloadType == g722PayloadType)
_timestampIncrement = g722RtpTimeincrement;
else
@@ -85,7 +86,6 @@ void AudioRtpSession::updateSessionMedia (AudioCodec *audioCodec)
_debug ("AudioSymmetricRtpSession: Codec frame size: %d", frameSize);
_debug ("AudioSymmetricRtpSession: RTP timestamp increment: %d", _timestampIncrement);
-
if (payloadType == g722PayloadType) {
_debug ("AudioSymmetricRtpSession: Setting G722 payload format");
_queue->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, g722RtpClockRate));
@@ -118,7 +118,7 @@ void AudioRtpSession::setSessionMedia (AudioCodec *audioCodec)
int smplRate = getCodecSampleRate();
bool dynamic = getHasDynamicPayload();
- // G722 requires timestamp to be incremented at 8 kHz
+ // G722 requires timestamp to be incremented at 8kHz
if (payloadType == g722PayloadType)
_timestampIncrement = g722RtpTimeincrement;
else
@@ -203,7 +203,7 @@ void AudioRtpSession::receiveSpeakerData ()
// DTMF over RTP, size must be over 4 in order to process it as voice data
if (size > 4)
- processDataDecode (spkrDataIn, size);
+ processDataDecode (spkrDataIn, size, adu->getType());
delete adu;
}
diff --git a/sflphone-common/src/audio/codecs/alaw.cpp b/sflphone-common/src/audio/codecs/alaw.cpp
index 7f488952d154e119ce20130a078108fd1a1e6ee4..db03155fd9bd4688533816ec7f8f86819edc045f 100644
--- a/sflphone-common/src/audio/codecs/alaw.cpp
+++ b/sflphone-common/src/audio/codecs/alaw.cpp
@@ -29,8 +29,10 @@
* as that of the covered work.
*/
+#include "global.h"
#include "../common.h"
#include "audiocodec.h"
+#include <cassert>
class Alaw : public sfl::AudioCodec
{
@@ -48,23 +50,24 @@ class Alaw : public sfl::AudioCodec
virtual ~Alaw() {}
- virtual int decode (short *dst, unsigned char *src, unsigned int size) {
- int16* end = dst+size;
+ virtual int decode (short *dst, unsigned char *src, size_t buf_size) {
+ assert(buf_size == _frameSize / 2 /* compression factor = 2:1 */ * sizeof(SFLDataFormat));
+ unsigned char* end = src+buf_size;
- while (dst<end)
+ while (src<end)
*dst++ = ALawDecode (*src++);
- return size<<1;
+ return _frameSize;
}
- virtual int encode (unsigned char *dst, short *src, unsigned int size) {
- size >>= 1;
- uint8* end = dst+size;
+ virtual int encode (unsigned char *dst, short *src, size_t buf_size) {
+ assert(buf_size >= _frameSize / 2 /* compression factor = 2:1 */ * sizeof(SFLDataFormat));
+ uint8* end = dst+_frameSize;
while (dst<end)
*dst++ = ALawEncode (*src++);
- return size;
+ return _frameSize / 2 /* compression factor = 2:1 */ * sizeof(SFLDataFormat);
}
diff --git a/sflphone-common/src/audio/codecs/audiocodec.h b/sflphone-common/src/audio/codecs/audiocodec.h
index aa05c008e4f2d2a532d863168de087e93dee07f5..b2d23fe0026bb111af67d18ef076de60cf57e159 100644
--- a/sflphone-common/src/audio/codecs/audiocodec.h
+++ b/sflphone-common/src/audio/codecs/audiocodec.h
@@ -91,15 +91,17 @@ class AudioCodec : public Codec
/**
* Decode an input buffer and fill the output buffer with the decoded data
- * @return the number of bytes decoded
+ * @param buffer_size : the size of the input buffer
+ * @return the number of samples decoded
*/
- virtual int decode (short *, unsigned char *, unsigned int) = 0;
+ virtual int decode (short *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 *, short *, unsigned int) = 0;
+ virtual int encode (unsigned char *dst, short *src, size_t buffer_size) = 0;
/**
* @Override
diff --git a/sflphone-common/src/audio/codecs/celtcodec.cpp b/sflphone-common/src/audio/codecs/celtcodec.cpp
index 9e9f15141f9b866b6aadabd8f175e8171b8d8b4a..1d75bf28d5a871df8b44a2ea4c42bc305e21e051 100644
--- a/sflphone-common/src/audio/codecs/celtcodec.cpp
+++ b/sflphone-common/src/audio/codecs/celtcodec.cpp
@@ -46,7 +46,6 @@ class Celt : public sfl::AudioCodec
_hasDynamicPayload = true;
initCelt();
-
}
Celt (const Celt&);
@@ -124,27 +123,26 @@ class Celt : public sfl::AudioCodec
celt_mode_destroy (_mode);
}
- virtual int decode (short *dst, unsigned char *src, unsigned int size) {
+ virtual int decode (short *dst, unsigned char *src, size_t buf_size) {
#ifdef BUILD_CELT_91 // == 91
//int err = 0;
- /*err =*/ celt_decode (_dec, src, size, (celt_int16*) dst, size);
+ /*err =*/ celt_decode (_dec, src, buf_size, (celt_int16*) dst, _frameSize);
#endif
#ifdef BUILD_CELT_71
//int err = 0; // FIXME: check error code
- /*err =*/ celt_decode (_dec, src, size, (celt_int16*) dst);
+ /*err =*/ celt_decode (_dec, src, buf_size, (celt_int16*) dst);
#endif
- return _frameSize * sizeof (celt_int16);
+ return _frameSize;
}
- virtual int encode (unsigned char *dst, short *src, unsigned int size) {
+ virtual int encode (unsigned char *dst, short *src, size_t buf_size) {
int len = 0;
#ifdef BUILD_CELT_91// == 91
- len = celt_encode (_enc, (celt_int16*) src, size, dst, 40);
+ len = celt_encode (_enc, (celt_int16*) src, _frameSize, dst, buf_size);
#endif
#ifdef BUILD_CELT_71
- len = celt_encode (_enc, (celt_int16*) src, (celt_int16 *) src, dst, 40);
+ len = celt_encode (_enc, (celt_int16*) src, (celt_int16 *) src, dst, buf_size);
#endif
- // returns the number of bytes writen
return len;
}
diff --git a/sflphone-common/src/audio/codecs/g722.cpp b/sflphone-common/src/audio/codecs/g722.cpp
index 4ea5e7c86950ad9c6b165d2e922dc258352c777e..26edcb1dfd884d497f5edee9df5777bd8be70993 100644
--- a/sflphone-common/src/audio/codecs/g722.cpp
+++ b/sflphone-common/src/audio/codecs/g722.cpp
@@ -32,12 +32,14 @@
+#include "global.h"
#include "../common.h"
#include "audiocodec.h"
#include "g722.h"
#include <stdlib.h>
#include <string.h>
+#include <cassert>
#define TRUE 1
#define FALSE 0
@@ -70,25 +72,15 @@ class G722 : public sfl::AudioCodec
g722_encode_release();
}
- virtual int decode (short *dst, unsigned char *src, unsigned int size) {
-
- int in_byte = size;
- int out_samples;
-
- out_samples = g722_decode ( (int16_t*) dst, (const uint8_t*) src, in_byte);
-
- return out_samples * 2;
+ virtual int decode (short *dst, unsigned char *src, size_t buf_size) {
+ assert(buf_size == _frameSize / sizeof(SFLDataFormat) * encode_s->bits_per_sample / 8);
+ return g722_decode ( (int16_t*) dst, (const uint8_t*) src, buf_size);
}
- virtual int encode (unsigned char *dst, short *src, unsigned int size) {
-
- // 2 bytes per sample (int16)
- int in_samples = size / 2;
- int out_bytes;
-
- out_bytes = g722_encode ( (uint8_t*) dst, (const int16_t*) src, in_samples);
-
- return out_bytes;
+ virtual int encode (unsigned char *dst, short *src, size_t buf_size) {
+ int out = g722_encode ( (uint8_t*) dst, (const int16_t*) src, _frameSize);
+ assert((size_t)out <= buf_size);
+ return out;
}
diff --git a/sflphone-common/src/audio/codecs/gsmcodec.cpp b/sflphone-common/src/audio/codecs/gsmcodec.cpp
index 0cba124532f275d92e4ec344cc2d887365816960..a1033cb639eaa742e65d473a58ad8cbde4dc0484 100644
--- a/sflphone-common/src/audio/codecs/gsmcodec.cpp
+++ b/sflphone-common/src/audio/codecs/gsmcodec.cpp
@@ -33,6 +33,8 @@
#include "audiocodec.h"
extern "C" {
#include <gsm/gsm.h>
+#include <cassert>
+
}
/**
@@ -52,10 +54,10 @@ class Gsm : public sfl::AudioCodec
_hasDynamicPayload = false;
if (! (_decode_gsmhandle = gsm_create()))
- printf ("ERROR: decode_gsm_create");
+ printf ("ERROR: decode_gsm_create\n");
if (! (_encode_gsmhandle = gsm_create()))
- printf ("AudioCodec: ERROR: encode_gsm_create");
+ printf ("AudioCodec: ERROR: encode_gsm_create\n");
}
Gsm (const Gsm&);
@@ -67,20 +69,19 @@ class Gsm : public sfl::AudioCodec
gsm_destroy (_encode_gsmhandle);
}
- virtual int decode (short * dst, unsigned char * src, unsigned int size) {
- // _debug("Decoded by gsm ");
- (void) size;
+ virtual int decode (short * dst, unsigned char * src, size_t buf_size) {
+ assert(buf_size == 33);
+ (void) buf_size;
if (gsm_decode (_decode_gsmhandle, (gsm_byte*) src, (gsm_signal*) dst) < 0)
- printf ("ERROR: gsm_decode");
+ printf ("ERROR: gsm_decode\n");
- return 320;
+ return _frameSize;
}
- virtual int encode (unsigned char * dst, short * src, unsigned int size) {
-
- // _debug("Encoded by gsm ");
- (void) size;
+ virtual int encode (unsigned char * dst, short * src, size_t buf_size) {
+ (void) buf_size;
+ assert(buf_size >= 33);
gsm_encode (_encode_gsmhandle, (gsm_signal*) src, (gsm_byte*) dst);
return 33;
}
diff --git a/sflphone-common/src/audio/codecs/speexcodec_nb.cpp b/sflphone-common/src/audio/codecs/speexcodec_nb.cpp
index 151b781b625ff8185f88bb7982b9b3d9b579e1eb..2225e2c71e85f6bf2e9359f637ee4359a061fd8c 100644
--- a/sflphone-common/src/audio/codecs/speexcodec_nb.cpp
+++ b/sflphone-common/src/audio/codecs/speexcodec_nb.cpp
@@ -94,23 +94,16 @@ class Speex : public sfl::AudioCodec
_speex_enc_state = 0;
}
- virtual int decode (short *dst, unsigned char *src, unsigned int size) {
-
- // int ratio = 320 / _speex_frame_size;
- speex_bits_read_from (&_speex_dec_bits, (char*) src, size);
+ virtual int decode (short *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 size in bytes
- return _frameSize * 2;
+ return _frameSize;
}
- virtual int encode (unsigned char *dst, short *src, unsigned int size) {
+ virtual int encode (unsigned char *dst, short *src, size_t buf_size) {
speex_bits_reset (&_speex_enc_bits);
-
speex_encode_int (_speex_enc_state, src, &_speex_enc_bits);
- int nbBytes = speex_bits_write (&_speex_enc_bits, (char*) dst, size);
-
- return nbBytes;
+ return speex_bits_write (&_speex_enc_bits, (char*) dst, buf_size);
}
private:
diff --git a/sflphone-common/src/audio/codecs/speexcodec_ub.cpp b/sflphone-common/src/audio/codecs/speexcodec_ub.cpp
index b3f956297d4d14e525217132c1ec841c33d5d8f2..d22adee0eab93432622c9646e2487b496e47fa37 100644
--- a/sflphone-common/src/audio/codecs/speexcodec_ub.cpp
+++ b/sflphone-common/src/audio/codecs/speexcodec_ub.cpp
@@ -95,24 +95,17 @@ class Speex : public sfl::AudioCodec
}
- virtual int decode (short *dst, unsigned char *src, unsigned int size) {
-
- // int ratio = 320 / _speex_frame_size;
- speex_bits_read_from (&_speex_dec_bits, (char*) src, size);
+ virtual int decode (short *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 the nuber of byte, not the number of sample
- return _frameSize * 2;
+ return _frameSize;
}
- virtual int encode (unsigned char *dst, short *src, unsigned int size) {
+ virtual int encode (unsigned char *dst, short *src, size_t buf_size) {
speex_bits_reset (&_speex_enc_bits);
-
speex_encode_int (_speex_enc_state, src, &_speex_enc_bits);
speex_bits_nbytes (&_speex_enc_bits);
- int nbBytes = speex_bits_write (&_speex_enc_bits, (char*) dst, size);
-
- return nbBytes;
+ return speex_bits_write (&_speex_enc_bits, (char*) dst, buf_size);
}
private:
diff --git a/sflphone-common/src/audio/codecs/speexcodec_wb.cpp b/sflphone-common/src/audio/codecs/speexcodec_wb.cpp
index 482d9f47cf4fd0f21b9553b6c9a95b9d43545609..04199c4372b65069408e5e72f476d63f611fd903 100644
--- a/sflphone-common/src/audio/codecs/speexcodec_wb.cpp
+++ b/sflphone-common/src/audio/codecs/speexcodec_wb.cpp
@@ -93,25 +93,16 @@ class Speex : public sfl::AudioCodec
}
- virtual int decode (short *dst, unsigned char *src, unsigned int size) {
-
- // int ratio = 320 / _speex_frame_size;
-
- speex_bits_read_from (&_speex_dec_bits, (char*) src, size);
+ virtual int decode (short *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 size in bytes
- return _frameSize * 2;
+ return _frameSize;
}
- virtual int encode (unsigned char *dst, short *src, unsigned int size) {
+ virtual int encode (unsigned char *dst, short *src, size_t buf_size) {
speex_bits_reset (&_speex_enc_bits);
-
- //printf ("Codec::codecEncode() size %i\n", size);
speex_encode_int (_speex_enc_state, src, &_speex_enc_bits);
- int nbBytes = speex_bits_write (&_speex_enc_bits, (char*) dst, size);
- //printf ("Codec::codecEncode() nbBytes %i\n", nbBytes);
- return nbBytes;
+ return speex_bits_write (&_speex_enc_bits, (char*) dst, buf_size);
}
private:
diff --git a/sflphone-common/src/audio/codecs/ulaw.cpp b/sflphone-common/src/audio/codecs/ulaw.cpp
index 0032eb9dc9282f5ad7934bd6b47861a2dd29793c..4faf466a16a804b56bdad3a255d44e1491c06865 100644
--- a/sflphone-common/src/audio/codecs/ulaw.cpp
+++ b/sflphone-common/src/audio/codecs/ulaw.cpp
@@ -30,13 +30,13 @@
*/
-
+#include "global.h"
#include "../common.h"
#include "audiocodec.h"
+#include <cassert>
class Ulaw : public sfl::AudioCodec
{
-
public:
// 0 PCMU A 8000 1 [RFC3551]
Ulaw (int payload=0)
@@ -48,23 +48,24 @@ class Ulaw : public sfl::AudioCodec
_hasDynamicPayload = false;
}
- virtual int decode (short *dst, unsigned char *src, unsigned int size) {
- int16* end = dst+size;
+ virtual int decode (short *dst, unsigned char *src, size_t buf_size) {
+ assert(buf_size == _frameSize / 2 /* compression factor = 2:1 */ * sizeof(SFLDataFormat));
+ unsigned char* end = src+buf_size;
- while (dst<end)
+ while (src<end)
*dst++ = ULawDecode (*src++);
- return size<<1;
+ return _frameSize;
}
- virtual int encode (unsigned char *dst, short *src, unsigned int size) {
- size >>= 1;
- uint8* end = dst+size;
+ virtual int encode (unsigned char *dst, short *src, size_t buf_size) {
+ assert(buf_size >= _frameSize / 2 /* compression factor = 2:1 */ * sizeof(SFLDataFormat));
+ uint8* end = dst+_frameSize;
while (dst<end)
*dst++ = ULawEncode (*src++);
- return size;
+ return _frameSize / 2 /* compression factor = 2:1 */ * sizeof(SFLDataFormat);;
}
int ULawDecode (uint8 ulaw) {
diff --git a/sflphone-common/src/audio/samplerateconverter.cpp b/sflphone-common/src/audio/samplerateconverter.cpp
index ca4e54def666fd2ef60f2a90695b2b55afef214b..8bfcbeb68756a7c16f0aec2c85ea6467a9692c3c 100644
--- a/sflphone-common/src/audio/samplerateconverter.cpp
+++ b/sflphone-common/src/audio/samplerateconverter.cpp
@@ -64,16 +64,17 @@ SamplerateConverter::Short2FloatArray (const short *in, float *out, int len)
//TODO Add ifdef for int16 or float32 type
void SamplerateConverter::resample (SFLDataFormat* dataIn , SFLDataFormat* dataOut , int inputFreq , int outputFreq , int nbSamples)
{
- assert(outputFreq <= _maxFreq);
-
double sampleFactor = (double) outputFreq / inputFreq;
if (sampleFactor == 1.0)
return;
- int outSamples = nbSamples * sampleFactor;
- if (outSamples > _samples) {
+ unsigned int outSamples = nbSamples * sampleFactor;
+ unsigned int maxSamples = outSamples;
+ if (maxSamples < (unsigned int)nbSamples)
+ maxSamples = nbSamples;
+ if (maxSamples > _samples) {
/* grow buffer if needed */
- _samples = outSamples;
+ _samples = maxSamples;
delete [] _floatBufferIn;
delete [] _floatBufferOut;
_floatBufferIn = new float32[_samples];
diff --git a/sflphone-common/src/audio/sound/audiofile.cpp b/sflphone-common/src/audio/sound/audiofile.cpp
index 9c8d4d25ddc41425fd5a700131b4499349bf00db..619be1aec84eb72b970e75c0adb3ae5eba77aaab 100644
--- a/sflphone-common/src/audio/sound/audiofile.cpp
+++ b/sflphone-common/src/audio/sound/audiofile.cpp
@@ -43,100 +43,81 @@
#include "manager.h"
-// load file in mono format
RawFile::RawFile(const std::string& name, sfl::AudioCodec* codec, unsigned int sampleRate)
: audioCodec (codec)
{
filepath = name;
-
- // no filename to load
if (filepath.empty())
throw AudioFileException("Unable to open audio file: filename is empty");
- std::fstream file;
+ std::fstream file;
file.open (filepath.c_str(), std::fstream::in);
- if (!file.is_open()) {
+ if (!file.is_open())
throw AudioFileException("Unable to open audio file");
- }
- // get length of file:
file.seekg (0, std::ios::end);
- int length = file.tellg();
+ size_t length = file.tellg();
file.seekg (0, std::ios::beg);
- // allocate memory:
- char fileBuffer[length];
-
- // read data as a block:
+ char *fileBuffer = new char[length];
file.read (fileBuffer,length);
-
file.close();
-
- // Decode file.ul
- // expandedsize is the number of bytes, not the number of int
- // expandedsize should be exactly two time more, else failed
- int16 monoBuffer[length];
-
- unsigned int expandedsize = audioCodec->decode (monoBuffer, reinterpret_cast<unsigned char *>(fileBuffer), length);
- if (expandedsize != length * sizeof(int16)) {
- throw AudioFileException("Audio file error on loading audio file!");
+ unsigned int frameSize = audioCodec->getFrameSize();
+ unsigned int bitrate = audioCodec->getBitRate() * 1000 / 8;
+ unsigned int audioRate = audioCodec->getClockRate();
+ unsigned int encFrameSize = frameSize * bitrate / audioRate;
+ unsigned int decodedSize = length * (frameSize / encFrameSize);
+
+ SFLDataFormat *monoBuffer = new SFLDataFormat[decodedSize];
+ SFLDataFormat *bufpos = monoBuffer;
+ unsigned char *filepos = reinterpret_cast<unsigned char *>(fileBuffer);
+ _size = decodedSize;
+ while(length >= encFrameSize) {
+ bufpos += audioCodec->decode (bufpos, filepos, encFrameSize);
+ filepos += encFrameSize;
+ length -= encFrameSize;
}
+ delete[] fileBuffer;
- unsigned int nbSampling = expandedsize / sizeof(int16);
-
- // we need to change the sample rating here:
- // case 1: we don't have to resample : only do splitting and convert
- if (sampleRate == 8000) {
- // just s
- _size = nbSampling;
- _buffer = new SFLDataFormat[_size];
+ if (sampleRate == audioRate) {
#ifdef DATAFORMAT_IS_FLOAT
- // src to dest
- src_short_to_float_array (monoBuffer, _buffer, nbSampling);
+ _buffer = new SFLDataFormat[_size];
+ src_short_to_float_array (monoBuffer, _buffer, _size);
+ delete[] monoBuffer;
#else
- // dest to src
- memcpy (_buffer, monoBuffer, _size*sizeof (SFLDataFormat));
+ _buffer = monoBuffer;
#endif
} else {
- // case 2: we need to convert it and split it
- // convert here
- double factord = (double) sampleRate / 8000;
- float* floatBufferIn = new float[nbSampling];
- int sizeOut = (int) (ceil (factord*nbSampling));
- src_short_to_float_array (monoBuffer, floatBufferIn, nbSampling);
- SFLDataFormat* bufferTmp = new SFLDataFormat[sizeOut];
+ 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];
SRC_DATA src_data;
src_data.data_in = floatBufferIn;
- src_data.input_frames = nbSampling;
+ src_data.input_frames = _size;
src_data.output_frames = sizeOut;
src_data.src_ratio = factord;
#ifdef DATAFORMAT_IS_FLOAT
- // case number 2.1: the output is float32 : convert directly in _bufferTmp
- src_data.data_out = bufferTmp;
+ src_data.data_out = _buffer;
src_simple (&src_data, SRC_SINC_BEST_QUALITY, 1);
#else
- // case number 2.2: the output is int16 : convert and change to int16
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, bufferTmp, src_data.output_frames_gen);
+ src_float_to_short_array (floatBufferOut, _buffer, src_data.output_frames_gen);
delete [] floatBufferOut;
#endif
delete [] floatBufferIn;
- nbSampling = src_data.output_frames_gen;
-
- // if we are in mono, we send the bufferTmp location and don't delete it
- // else we split the audio in 2 and put it into buffer
- _size = nbSampling;
- _buffer = bufferTmp; // just send the buffer pointer;
- bufferTmp = 0;
+ _size = src_data.output_frames_gen;
}
}
@@ -211,7 +192,7 @@ WaveFile::WaveFile (const std::string& fileName, unsigned int audioSamplingRate)
fileStream.read (data, 4);
// Sample rate converter initialized with 88200 sample long
- int converterSamples = (srate > audioSamplingRate) ? srate : audioSamplingRate;
+ int converterSamples = ((unsigned int)srate > audioSamplingRate) ? srate : audioSamplingRate;
SamplerateConverter _converter (converterSamples);
// Get length of data from the header.
@@ -240,7 +221,7 @@ WaveFile::WaveFile (const std::string& fileName, unsigned int audioSamplingRate)
nbSamples /= 2;
}
- if (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);
diff --git a/sflphone-common/src/iax/iaxvoiplink.cpp b/sflphone-common/src/iax/iaxvoiplink.cpp
index 9c597dfb5652766a861025286a3d26c755fd2577..f4ca30bdcfa7a211df0edca4aecd19dfa64023af 100644
--- a/sflphone-common/src/iax/iaxvoiplink.cpp
+++ b/sflphone-common/src/iax/iaxvoiplink.cpp
@@ -289,14 +289,15 @@ IAXVoIPLink::sendAudioFromMic (void)
int compSize;
unsigned int audioRate = audioCodec->getClockRate();
int outSamples;
+ SFLDataFormat *in;
if (audioRate != mainBufferSampleRate) {
- int outSamples = ((float) samples * ( (float) mainBufferSampleRate / (float) audioRate));
converter->resample (decData , resampledData , audioRate, mainBufferSampleRate, samples);
- compSize = audioCodec->encode (encodedData, resampledData , outSamples * sizeof(SFLDataFormat));
+ in = resampledData;
} else {
outSamples = samples;
- compSize = audioCodec->encode (encodedData, decData, bytes);
+ in = decData;
}
+ compSize = audioCodec->encode (encodedData, in, DEC_BUFFER_SIZE);
// Send it out!
_mutexIAX.enterMutex();
@@ -915,15 +916,15 @@ IAXVoIPLink::iaxHandleVoiceEvent (iax_event* event, IAXCall* call)
size = max;
}
- int expandedSize = audioCodec->decode (decData, data , size);
+ int samples = audioCodec->decode (decData, data , size);
+ int outSize = samples * sizeof(SFLDataFormat);
unsigned int audioRate = audioCodec->getClockRate();
- int inSamples = expandedSize / sizeof(SFLDataFormat);
if (audioRate != mainBufferSampleRate) {
- int outSize = expandedSize * (mainBufferSampleRate / audioRate);
- converter->resample (decData, resampledData, mainBufferSampleRate, audioRate, inSamples);
+ outSize = (double)outSize * (mainBufferSampleRate / audioRate);
+ converter->resample (decData, resampledData, mainBufferSampleRate, audioRate, samples);
audiolayer->getMainBuffer()->putData (resampledData, outSize, call->getCallId());
} else {
- audiolayer->getMainBuffer()->putData (decData, expandedSize, call->getCallId());
+ audiolayer->getMainBuffer()->putData (decData, outSize, call->getCallId());
}
}