diff --git a/configure.ac b/configure.ac
index 8b41bbad0..95386d296 100644
--- a/configure.ac
+++ b/configure.ac
@@ -11,6 +11,7 @@ AC_SUBST(MP4FF_LIB)
AC_SUBST(MP4FF_SUBDIR)
AC_PROG_CC
+AM_PROG_CC_C_O
AC_PROG_INSTALL
AC_PROG_LIBTOOL
AC_PROG_MAKE_SET
@@ -79,6 +80,7 @@ AC_ARG_ENABLE(audiofile,[ --disable-audiofile disable audiofile support, di
AC_ARG_ENABLE(mod,[ --enable-mod enable MOD support (default: disable],[enable_mod=$enableval],[enable_mod=yes])
AC_ARG_ENABLE(mpc,[ --disable-mpc disable musepack (MPC) support (default: enable)],[enable_mpc=$enableval],[enable_mpc=yes])
AC_ARG_ENABLE(id3,[ --disable-id3 disable id3 support (default: enable)],[enable_id3=$enableval],[enable_id3=yes])
+AC_ARG_ENABLE(lsr,[ --disable-lsr disable libsamplerate support (default: enable)],[enable_lsr=$enableval],[enable_lsr=yes])
AC_ARG_WITH(tremor,[[ --with-tremor[=PFX] Use Tremor(vorbisidec) integer Ogg-Vorbis decoder (with optional prefix)]], use_tremor=yes; test x$withval != xyes && tremor_prefix="$withval",)
AC_ARG_WITH(tremor-libraries,[ --with-tremor-libraries=DIR Directory where Tremor library is installed (optional)], tremor_libraries="$withval", tremor_libraries="")
@@ -101,6 +103,10 @@ AC_ARG_WITH(faad-libraries,[ --with-faad-libraries=DIR Directory where faad2
AC_ARG_WITH(faad-includes,[ --with-faad-includes=DIR Directory where faad2 header files are installed (optional)], faad_includes="$withval", faad_includes="")
AC_ARG_WITH(zeroconf,[[ --with-zeroconf=[auto|avahi|bonjour|no] Enable zeroconf backend (default=auto)]], with_zeroconf="$withval", with_zeroconf="auto")
+AC_ARG_WITH(lsr,[ --with-src=PFX Prefix where libsamplerate is installed], src_prefix="$withval", src_prefix="")
+AC_ARG_WITH(lsr-libraries,[ --with-lsr-libraries=DIR Directory where libsamplerate library is installed (optional)], lsr_libraries="$withval", lsr_libraries="")
+AC_ARG_WITH(lsr-includes,[ --with-lsr-includes=DIR Directory where libsamplerate header files are installed (optional)], lsr_includes="$withval", lsr_includes="")
+
AC_C_BIGENDIAN
AC_CHECK_SIZEOF(short)
@@ -186,6 +192,12 @@ if test x$enable_pulse = xyes; then
[enable_pulse=no;AC_MSG_WARN([PulseAudio not found -- disabling])])
fi
+if test x$enable_lsr = xyes; then
+ PKG_CHECK_MODULES([SAMPLERATE], [samplerate >= 0.0.15],
+ [enable_lsr=yes;AC_DEFINE([HAVE_LIBSAMPLERATE], 1, [Define to enable libsamplerate])] MPD_LIBS="$MPD_LIBS $SAMPLERATE_LIBS" MPD_CFLAGS="$MPD_CFLAGS $SAMPLERATE_CFLAGS",
+ [enable_lsr=no;AC_MSG_WARN([libsamplerate not found -- disabling])])
+fi
+
if test x$enable_mvp = xyes; then
AC_DEFINE(HAVE_MVP,1,[Define to enable Hauppauge Media MVP support])
fi
@@ -777,8 +789,15 @@ if
fi
echo ""
-
echo " Other features:"
+
+if test x$enable_lsr = xyes; then
+ echo " libsamplerate support .........enabled"
+else
+ echo " libsamplerate support .........disabled"
+fi
+
+
if test x$with_zeroconf != xno; then
echo " Zeroconf support ..............$with_zeroconf"
else
diff --git a/doc/mpd.conf.5 b/doc/mpd.conf.5
index 821500e8b..572738234 100644
--- a/doc/mpd.conf.5
+++ b/doc/mpd.conf.5
@@ -126,6 +126,35 @@ This is the gain (in dB) applied to songs with ReplayGain tags.
.B volume_normalization <yes or no>
If yes, mpd will normalize the volume of songs as they play. Default is no.
.TP
+.B samplerate_converter <integer or prefix>
+Specifies the libsamplerate converter to use.
+The supplied value should either be an integer or a prefix of the name of a converter.
+The list of available converters at the time of writing is below.
+More converters may exist, consult the
+documentation of the Secret Rabbit Code libsamplerate (at http://www.mega-nerd.com/SRC/) for details.
+.RS
+.HP
+Best Sinc Interpolator (0)
+
+Band limited sinc interpolation, best quality, 97dB SNR, 96% BW.
+.HP
+Medium Sinc Interpolator (1)
+
+Band limited sinc interpolation, medium quality, 97dB SNR, 90% BW.
+.HP
+Fastest Sinc Interpolator (2, default)
+
+Band limited sinc interpolation, fastest, 97dB SNR, 80% BW.
+.HP
+ZOH Interpolator (3)
+
+Zero order hold interpolator, very fast, very poor quality with audible distortions.
+.HP
+Linear Interpolator (4)
+
+Linear interpolator, very fast, poor quality.
+.RE
+.TP
.B audio_buffer_size <size in KiB>
This specifies the size of the audio output buffer that mpd uses. The default
is 2048.
diff --git a/doc/mpdconf.example b/doc/mpdconf.example
index c7a294214..20ab89ec4 100644
--- a/doc/mpdconf.example
+++ b/doc/mpdconf.example
@@ -123,6 +123,12 @@ error_file "~/.mpd/mpd.error"
#
#audio_output_format "44100:16:2"
#
+# Specifies the libsamplerate converter to use (if compiled in),
+# see man 5 mpd.conf for more information.
+#
+#samplerate_converter <integer or prefix>
+#
+
################################################################
diff --git a/src/conf.c b/src/conf.c
index 0960dfbbc..7dfea6327 100644
--- a/src/conf.c
+++ b/src/conf.c
@@ -157,6 +157,7 @@ void initConf(void)
registerConfigParam(CONF_REPLAYGAIN, 0, 0);
registerConfigParam(CONF_REPLAYGAIN_PREAMP, 0, 0);
registerConfigParam(CONF_VOLUME_NORMALIZATION, 0, 0);
+ registerConfigParam(CONF_SAMPLERATE_CONVERTER, 0, 0);
registerConfigParam(CONF_AUDIO_BUFFER_SIZE, 0, 0);
registerConfigParam(CONF_BUFFER_BEFORE_PLAY, 0, 0);
registerConfigParam(CONF_HTTP_BUFFER_SIZE, 0, 0);
diff --git a/src/conf.h b/src/conf.h
index e2f568525..1cd4cf322 100644
--- a/src/conf.h
+++ b/src/conf.h
@@ -43,6 +43,7 @@
#define CONF_REPLAYGAIN "replaygain"
#define CONF_REPLAYGAIN_PREAMP "replaygain_preamp"
#define CONF_VOLUME_NORMALIZATION "volume_normalization"
+#define CONF_SAMPLERATE_CONVERTER "samplerate_converter"
#define CONF_AUDIO_BUFFER_SIZE "audio_buffer_size"
#define CONF_BUFFER_BEFORE_PLAY "buffer_before_play"
#define CONF_HTTP_BUFFER_SIZE "http_buffer_size"
diff --git a/src/pcm_utils.c b/src/pcm_utils.c
index b56ed33ac..f6b308258 100644
--- a/src/pcm_utils.c
+++ b/src/pcm_utils.c
@@ -21,11 +21,16 @@
#include "mpd_types.h"
#include "log.h"
#include "utils.h"
+#include "conf.h"
#include <string.h>
#include <math.h>
#include <assert.h>
+#ifdef HAVE_LIBSAMPLERATE
+#include <samplerate.h>
+#endif
+
void pcm_volumeChange(char *buffer, int bufferSize, AudioFormat * format,
int volume)
{
@@ -46,6 +51,9 @@ void pcm_volumeChange(char *buffer, int bufferSize, AudioFormat * format,
while (bufferSize > 0) {
temp32 = *buffer16;
temp32 *= volume;
+ temp32 += rand() & 511;
+ temp32 -= rand() & 511;
+ temp32 += 500;
temp32 /= 1000;
*buffer16 = temp32 > 32767 ? 32767 :
(temp32 < -32768 ? -32768 : temp32);
@@ -57,6 +65,9 @@ void pcm_volumeChange(char *buffer, int bufferSize, AudioFormat * format,
while (bufferSize > 0) {
temp32 = *buffer8;
temp32 *= volume;
+ temp32 += rand() & 511;
+ temp32 -= rand() & 511;
+ temp32 += 500;
temp32 /= 1000;
*buffer8 = temp32 > 127 ? 127 :
(temp32 < -128 ? -128 : temp32);
@@ -86,7 +97,11 @@ static void pcm_add(char *buffer1, char *buffer2, size_t bufferSize1,
while (bufferSize1 > 0 && bufferSize2 > 0) {
temp32 =
(vol1 * (*buffer16_1) +
- vol2 * (*buffer16_2)) / 1000;
+ vol2 * (*buffer16_2));
+ temp32 += rand() & 511;
+ temp32 -= rand() & 511;
+ temp32 += 500;
+ temp32 /= 1000;
*buffer16_1 =
temp32 > 32767 ? 32767 : (temp32 <
-32768 ? -32768 : temp32);
@@ -101,7 +116,11 @@ static void pcm_add(char *buffer1, char *buffer2, size_t bufferSize1,
case 8:
while (bufferSize1 > 0 && bufferSize2 > 0) {
temp32 =
- (vol1 * (*buffer8_1) + vol2 * (*buffer8_2)) / 1000;
+ (vol1 * (*buffer8_1) + vol2 * (*buffer8_2));
+ temp32 += rand() & 511;
+ temp32 -= rand() & 511;
+ temp32 += 500;
+ temp32 /= 1000;
*buffer8_1 =
temp32 > 127 ? 127 : (temp32 <
-128 ? -128 : temp32);
@@ -133,6 +152,38 @@ void pcm_mix(char *buffer1, char *buffer2, size_t bufferSize1,
format);
}
+#ifdef HAVE_LIBSAMPLERATE
+static int pcm_getSamplerateConverter(void) {
+ const char *conf, *test;
+ int convalgo = SRC_SINC_FASTEST;
+ int newalgo;
+ size_t len;
+
+ conf = getConfigParamValue(CONF_SAMPLERATE_CONVERTER);
+ if(conf) {
+ newalgo = strtol(conf, (char **)&test, 10);
+ if(*test) {
+ len = strlen(conf);
+ for(newalgo = 0; ; newalgo++) {
+ test = src_get_name(newalgo);
+ if(!test)
+ break; /* FAIL */
+ if(!strncasecmp(test, conf, len)) {
+ convalgo = newalgo;
+ break;
+ }
+ }
+ } else {
+ if(src_get_name(newalgo))
+ convalgo = newalgo;
+ /* else FAIL */
+ }
+ }
+ DEBUG("Selecting samplerate converter '%s'\n", src_get_name(convalgo));
+ return convalgo;
+}
+#endif
+
/* outFormat bits must be 16 and channels must be 2! */
void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, size_t
inSize, AudioFormat * outFormat, char *outBuffer)
@@ -234,6 +285,47 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, size_t
if (inFormat->sampleRate == outFormat->sampleRate) {
memcpy(outBuffer, dataChannelConv, dataChannelLen);
} else {
+#ifdef HAVE_LIBSAMPLERATE
+ static SRC_STATE *state = NULL;
+ static SRC_DATA data;
+ int error;
+ static double ratio = 0;
+ double newratio;
+
+ if(!state) {
+ state = src_new(pcm_getSamplerateConverter(), outFormat->channels, &error);
+ if(!state) {
+ ERROR("Cannot create new samplerate state: %s\n", src_strerror(error));
+ exit(EXIT_FAILURE);
+ } else {
+ DEBUG("Samplerate converter initialized\n");
+ }
+ }
+
+ newratio = (double)outFormat->sampleRate / (double)inFormat->sampleRate;
+ if(newratio != ratio) {
+ src_set_ratio(state, ratio = newratio);
+ DEBUG("Setting samplerate conversion ratio to %.2lf\n", ratio);
+ }
+
+ data.input_frames = dataChannelLen / 2 / outFormat->channels;
+ data.output_frames = pcm_sizeOfOutputBufferForAudioFormatConversion(inFormat, dataChannelLen, outFormat) / 2 / outFormat->channels;
+ data.src_ratio = (double)data.output_frames / (double)data.input_frames;
+
+ float conversionInBuffer[data.input_frames * outFormat->channels];
+ float conversionOutBuffer[data.output_frames * outFormat->channels];
+ data.data_in = conversionInBuffer;
+ data.data_out = conversionOutBuffer;
+
+ src_short_to_float_array((short *)dataChannelConv, data.data_in, data.input_frames * outFormat->channels);
+ error = src_process(state, &data);
+ if(error) {
+ ERROR("Cannot process samples: %s\n", src_strerror(error));
+ exit(EXIT_FAILURE);
+ }
+
+ src_float_to_short_array(data.data_out, (short *)outBuffer, data.output_frames * outFormat->channels);
+#else
/* only works if outFormat is 16-bit stereo! */
/* resampling code blatantly ripped from ESD */
mpd_uint32 rd_dat = 0;
@@ -241,11 +333,7 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, size_t
mpd_sint16 lsample, rsample;
mpd_sint16 *out = (mpd_sint16 *) outBuffer;
mpd_sint16 *in = (mpd_sint16 *) dataChannelConv;
- const int shift = sizeof(mpd_sint16) * outFormat->channels;
- mpd_uint32 nlen = (((dataChannelLen / shift) *
- (mpd_uint32) (outFormat->sampleRate)) /
- inFormat->sampleRate);
- nlen *= outFormat->channels;
+ mpd_uint32 nlen = pcm_sizeOfOutputBufferForAudioFormatConversion(inFormat, inSize, outFormat) / sizeof(mpd_sint16);
switch (outFormat->channels) {
case 1:
@@ -272,6 +360,7 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, size_t
}
break;
}
+#endif
}
return;
@@ -306,9 +395,9 @@ size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
}
}
- outSize = (((outSize / shift) * (mpd_uint32) (outFormat->sampleRate)) /
- inFormat->sampleRate);
-
+ outSize /= shift;
+ outSize = floor(0.5 + (double)outSize *
+ ((double)outFormat->sampleRate / (double)inFormat->sampleRate));
outSize *= shift;
return outSize;