From 42ae12dde963c3138d44064cec8b33a4cdb1b747 Mon Sep 17 00:00:00 2001
From: Alexandre Savard <alexandresavard@alexandresavard-desktop.(none)>
Date: Tue, 23 Feb 2010 14:30:13 -0500
Subject: [PATCH] [#2598] Use larger value for ALSA's period size and frame
size
---
sflphone-common/src/audio/alsa/alsalayer.cpp | 7 ++++---
sflphone-common/src/audio/pulseaudio/pulselayer.cpp | 3 ++-
sflphone-common/src/audio/samplerateconverter.cpp | 10 +++-------
3 files changed, 9 insertions(+), 11 deletions(-)
diff --git a/sflphone-common/src/audio/alsa/alsalayer.cpp b/sflphone-common/src/audio/alsa/alsalayer.cpp
index 920ab666b3..caa4d7f46c 100644
--- a/sflphone-common/src/audio/alsa/alsalayer.cpp
+++ b/sflphone-common/src/audio/alsa/alsalayer.cpp
@@ -135,7 +135,8 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize,
std::string pcmc = buildDeviceTopo (plugin, indexIn, 0);
- _converter = new SamplerateConverter (_audioSampleRate, _frameSize);
+ // use 1 sec buffer for resampling
+ _converter = new SamplerateConverter (_audioSampleRate, 1000);
// open_device (pcmp, pcmc, stream);
return true; // open_device (pcmp, pcmc, stream);
@@ -383,12 +384,12 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type, int rate)
int err;
int format;
int periods = 4;
- int periodsize = 1024;
+ int periodsize = 2048;
/* Allocate the snd_pcm_hw_params_t struct */
snd_pcm_hw_params_malloc (&hwparams);
- _periodSize = 940;
+ _periodSize = 2048;
/* Full configuration space */
if ( (err = snd_pcm_hw_params_any (pcm_handle, hwparams)) < 0) {
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
index 8da08d7c72..a9c03e2ed2 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -285,7 +285,8 @@ bool PulseLayer::openDevice (int indexIn UNUSED, int indexOut UNUSED, int sample
// _urgentRingBuffer.flushAll();
flushUrgent();
- _converter = new SamplerateConverter (_audioSampleRate, _frameSize*4);
+ // use 1 sec buffer for resampling
+ _converter = new SamplerateConverter (_audioSampleRate, 1000);
return true;
}
diff --git a/sflphone-common/src/audio/samplerateconverter.cpp b/sflphone-common/src/audio/samplerateconverter.cpp
index 9a2ace32e8..ba68515756 100644
--- a/sflphone-common/src/audio/samplerateconverter.cpp
+++ b/sflphone-common/src/audio/samplerateconverter.cpp
@@ -77,8 +77,7 @@ void SamplerateConverter::init (void)
_src_state_spkr = src_new (SRC_LINEAR, 1, &_src_err);
int nbSamplesMax = (int) (getFrequence() * getFramesize() / 1000);
- // TODO: fix this hack that make sure we have enought place in buffers to upsample
- nbSamplesMax = nbSamplesMax*4;
+
_floatBufferDownMic = new float32[nbSamplesMax];
_floatBufferUpMic = new float32[nbSamplesMax];
_floatBufferDownSpkr = new float32[nbSamplesMax];
@@ -104,9 +103,8 @@ int SamplerateConverter::upsampleData (SFLDataFormat* dataIn , SFLDataFormat* da
{
double upsampleFactor = (double) samplerate2 / samplerate1 ;
- //_debug("factor = %f" , upsampleFactor);
+
int nbSamplesMax = (int) (samplerate2 * getFramesize() / 1000);
- nbSamplesMax = nbSamplesMax*4;
if (upsampleFactor != 1 && dataIn != NULL) {
SRC_DATA src_data;
@@ -136,10 +134,8 @@ int SamplerateConverter::downsampleData (SFLDataFormat* dataIn , SFLDataFormat*
{
double downsampleFactor = (double) samplerate1 / samplerate2;
- //_debug("factor = %f" , downsampleFactor);
- int nbSamplesMax = (int) (samplerate1 * getFramesize() / 1000);
- nbSamplesMax = nbSamplesMax*4;
+ int nbSamplesMax = (int) (samplerate1 * getFramesize() / 1000);
if (downsampleFactor != 1) {
SRC_DATA src_data;
--
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