From 7cf8ce36dcb16444162415815b6295946dec1186 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Rafa=C3=ABl=20Carr=C3=A9?=
<rafael.carre@savoirfairelinux.com>
Date: Wed, 10 Aug 2011 16:15:19 -0400
Subject: [PATCH] resampler: grow internal buffers dynamically
also fix wrong assertion about the final number of samples
---
sflphone-common/src/audio/alsa/alsalayer.cpp | 2 +-
.../audio/audiortp/AudioRtpRecordHandler.cpp | 2 +-
.../src/audio/pulseaudio/pulselayer.cpp | 2 +-
.../src/audio/samplerateconverter.cpp | 34 +++++++++++--------
.../src/audio/samplerateconverter.h | 5 ++-
sflphone-common/src/audio/sound/audiofile.cpp | 2 +-
sflphone-common/src/iax/iaxvoiplink.cpp | 2 +-
7 files changed, 29 insertions(+), 20 deletions(-)
diff --git a/sflphone-common/src/audio/alsa/alsalayer.cpp b/sflphone-common/src/audio/alsa/alsalayer.cpp
index 9291dc1f27..a05eef1778 100644
--- a/sflphone-common/src/audio/alsa/alsalayer.cpp
+++ b/sflphone-common/src/audio/alsa/alsalayer.cpp
@@ -129,7 +129,7 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int indexRing, int sampleRate,
_audioThread = NULL;
// use 1 sec buffer for resampling
- _converter = new SamplerateConverter (_audioSampleRate, 1000);
+ _converter = new SamplerateConverter (_audioSampleRate);
AudioLayer::_dcblocker = new DcBlocker();
AudioLayer::_audiofilter = new AudioProcessing (static_cast<Algorithm *> (_dcblocker));
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
index e4c741de1b..cc94eccff0 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
@@ -190,7 +190,7 @@ void AudioRtpRecordHandler::initBuffers()
// (internal buffers initialized with maximal sampling rate and frame size)
int rate = getCodecSampleRate();
int fs = getCodecFrameSize();
- _audioRtpRecord._converter = new SamplerateConverter (rate, fs);
+ _audioRtpRecord._converter = new SamplerateConverter (rate);
int nbSamplesMax = (int) ( (getCodecSampleRate() * getCodecFrameSize() / 1000));
_audioRtpRecord._micData = new SFLDataFormat[nbSamplesMax];
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
index e7f0811976..9cd8269f3c 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -403,7 +403,7 @@ void PulseLayer::openDevice (int indexIn UNUSED, int indexOut UNUSED, int indexR
flushUrgent();
// use 1 sec buffer for resampling
- _converter = new SamplerateConverter (_audioSampleRate, 1000);
+ _converter = new SamplerateConverter (_audioSampleRate);
// Instantiate the algorithm
AudioLayer::_dcblocker = new DcBlocker();
diff --git a/sflphone-common/src/audio/samplerateconverter.cpp b/sflphone-common/src/audio/samplerateconverter.cpp
index 2ea2e275c9..411c20389c 100644
--- a/sflphone-common/src/audio/samplerateconverter.cpp
+++ b/sflphone-common/src/audio/samplerateconverter.cpp
@@ -30,16 +30,17 @@
#include "samplerateconverter.h"
#include "manager.h"
+#include <cassert>
-SamplerateConverter::SamplerateConverter (int freq , int fs)
+SamplerateConverter::SamplerateConverter (int freq) : _maxFreq(freq)
{
int err;
_src_state = src_new (SRC_LINEAR, 1, &err);
- int nbSamplesMax = (int) ( (freq * fs) / 1000);
+ _samples = (freq * 20) / 1000; // start with 20 ms buffers
- _floatBufferIn = new float32[nbSamplesMax];
- _floatBufferOut = new float32[nbSamplesMax];
+ _floatBufferIn = new float32[_samples];
+ _floatBufferOut = new float32[_samples];
}
SamplerateConverter::~SamplerateConverter (void)
@@ -61,29 +62,34 @@ 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 samplerate1 , int samplerate2 , int nbSamples)
+void SamplerateConverter::resample (SFLDataFormat* dataIn , SFLDataFormat* dataOut , int inputFreq , int outputFreq , int nbSamples)
{
- double sampleFactor;
- if (samplerate1 > samplerate2)
- sampleFactor = (double) samplerate1 / samplerate2;
- else
- sampleFactor = (double) samplerate2 / samplerate1;
+ assert(outputFreq <= _maxFreq);
+ double sampleFactor = (double) inputFreq / outputFreq;
if (sampleFactor == 1)
return;
+ int outSamples = nbSamples * sampleFactor;
+ if (outSamples > _samples) {
+ /* grow buffer if needed */
+ _samples = outSamples;
+ delete [] _floatBufferIn;
+ delete [] _floatBufferOut;
+ _floatBufferIn = new float32[_samples];
+ _floatBufferOut = new float32[_samples];
+ }
+
SRC_DATA src_data;
src_data.data_in = _floatBufferIn;
src_data.data_out = _floatBufferOut;
src_data.input_frames = nbSamples;
- src_data.output_frames = nbSamples;
+ src_data.output_frames = outSamples;
src_data.src_ratio = sampleFactor;
src_data.end_of_input = 0; // More data will come
Short2FloatArray (dataIn , _floatBufferIn, nbSamples);
src_process (_src_state, &src_data);
- assert(nbSamples == src_data.output_frames_gen);
-
- src_float_to_short_array (_floatBufferOut, dataOut , nbSamples);
+ src_float_to_short_array (_floatBufferOut, dataOut , outSamples);
}
diff --git a/sflphone-common/src/audio/samplerateconverter.h b/sflphone-common/src/audio/samplerateconverter.h
index 18f095d735..6d17b8d671 100644
--- a/sflphone-common/src/audio/samplerateconverter.h
+++ b/sflphone-common/src/audio/samplerateconverter.h
@@ -47,7 +47,7 @@ class SamplerateConverter
* internal buffer size. Converter must be reinitialized
* every time these parameters change
*/
- SamplerateConverter (int freq, int frameSize);
+ SamplerateConverter (int freq);
/** Destructor */
~SamplerateConverter (void);
@@ -77,8 +77,11 @@ class SamplerateConverter
// Assignment Operator
SamplerateConverter& operator= (const SamplerateConverter& rh);
+ /* temporary buffers */
float32* _floatBufferIn;
float32* _floatBufferOut;
+ size_t _samples; // size in samples of temporary buffers
+ int _maxFreq; // maximal output frequency
SRC_STATE* _src_state;
};
diff --git a/sflphone-common/src/audio/sound/audiofile.cpp b/sflphone-common/src/audio/sound/audiofile.cpp
index 69761cdab7..b10382945d 100644
--- a/sflphone-common/src/audio/sound/audiofile.cpp
+++ b/sflphone-common/src/audio/sound/audiofile.cpp
@@ -212,7 +212,7 @@ WaveFile::WaveFile (const std::string& fileName, unsigned int audioSamplingRate)
// Sample rate converter initialized with 88200 sample long
int converterSamples = (srate > audioSamplingRate) ? srate : audioSamplingRate;
- SamplerateConverter _converter (converterSamples, 2000);
+ SamplerateConverter _converter (converterSamples);
// Get length of data from the header.
SINT32 bytes;
diff --git a/sflphone-common/src/iax/iaxvoiplink.cpp b/sflphone-common/src/iax/iaxvoiplink.cpp
index e6f6212ce6..ba70c8628e 100644
--- a/sflphone-common/src/iax/iaxvoiplink.cpp
+++ b/sflphone-common/src/iax/iaxvoiplink.cpp
@@ -77,7 +77,7 @@ IAXVoIPLink::IAXVoIPLink (const std::string& accountID) : VoIPLink ()
// to get random number for RANDOM_PORT
srand (time (NULL));
- converter = new SamplerateConverter (44100, 20);
+ converter = new SamplerateConverter (44100);
// int nbSamplesMax = (int) (converter->getFrequence() * converter->getFramesize() / 1000);
int nbSamplesMax = (44100 * 20) / 1000;
--
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