Add libsamplerate support, old resampling is still an option, but this sounds much better for those who need it and don't want to use pulseaudio. Reviewed by shank/avuton.

git-svn-id: https://svn.musicpd.org/mpd/trunk@5316 09075e82-0dd4-0310-85a5-a0d7c8717e4f

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

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.

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>
+#
+
 ################################################################
 
 

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);

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"

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 = pcm_sizeOfOutputBufferForAudioFormatConversion(inFormat, inSize, outFormat) / sizeof(mpd_sint16);
-		mpd_uint32 nlen = (((dataChannelLen / shift) *
-				    (mpd_uint32) (outFormat->sampleRate)) /
-				   inFormat->sampleRate);
-		nlen *= outFormat->channels;
 
 		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)) /
+	outSize /=  shift;
-		   inFormat->sampleRate);
+	outSize = floor(0.5 + (double)outSize *
-
+		((double)outFormat->sampleRate / (double)inFormat->sampleRate));
 	outSize *= shift;
 
 	return outSize;