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
This commit is contained in:
Avuton Olrich
parent
96c5976ccc
commit
79ef8ba248
21
configure.ac
21
configure.ac
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@ -11,6 +11,7 @@ AC_SUBST(MP4FF_LIB)
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AC_SUBST(MP4FF_SUBDIR)
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AC_PROG_CC
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AM_PROG_CC_C_O
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AC_PROG_INSTALL
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AC_PROG_LIBTOOL
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AC_PROG_MAKE_SET
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@ -79,6 +80,7 @@ AC_ARG_ENABLE(audiofile,[ --disable-audiofile disable audiofile support, di
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AC_ARG_ENABLE(mod,[ --enable-mod enable MOD support (default: disable],[enable_mod=$enableval],[enable_mod=yes])
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AC_ARG_ENABLE(mpc,[ --disable-mpc disable musepack (MPC) support (default: enable)],[enable_mpc=$enableval],[enable_mpc=yes])
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AC_ARG_ENABLE(id3,[ --disable-id3 disable id3 support (default: enable)],[enable_id3=$enableval],[enable_id3=yes])
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AC_ARG_ENABLE(lsr,[ --disable-lsr disable libsamplerate support (default: enable)],[enable_lsr=$enableval],[enable_lsr=yes])
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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",)
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AC_ARG_WITH(tremor-libraries,[ --with-tremor-libraries=DIR Directory where Tremor library is installed (optional)], tremor_libraries="$withval", tremor_libraries="")
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@ -101,6 +103,10 @@ AC_ARG_WITH(faad-libraries,[ --with-faad-libraries=DIR Directory where faad2
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AC_ARG_WITH(faad-includes,[ --with-faad-includes=DIR Directory where faad2 header files are installed (optional)], faad_includes="$withval", faad_includes="")
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AC_ARG_WITH(zeroconf,[[ --with-zeroconf=[auto|avahi|bonjour|no] Enable zeroconf backend (default=auto)]], with_zeroconf="$withval", with_zeroconf="auto")
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AC_ARG_WITH(lsr,[ --with-src=PFX Prefix where libsamplerate is installed], src_prefix="$withval", src_prefix="")
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AC_ARG_WITH(lsr-libraries,[ --with-lsr-libraries=DIR Directory where libsamplerate library is installed (optional)], lsr_libraries="$withval", lsr_libraries="")
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AC_ARG_WITH(lsr-includes,[ --with-lsr-includes=DIR Directory where libsamplerate header files are installed (optional)], lsr_includes="$withval", lsr_includes="")
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AC_C_BIGENDIAN
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AC_CHECK_SIZEOF(short)
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@ -186,6 +192,12 @@ if test x$enable_pulse = xyes; then
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[enable_pulse=no;AC_MSG_WARN([PulseAudio not found -- disabling])])
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fi
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if test x$enable_lsr = xyes; then
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PKG_CHECK_MODULES([SAMPLERATE], [samplerate >= 0.0.15],
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[enable_lsr=yes;AC_DEFINE([HAVE_LIBSAMPLERATE], 1, [Define to enable libsamplerate])] MPD_LIBS="$MPD_LIBS $SAMPLERATE_LIBS" MPD_CFLAGS="$MPD_CFLAGS $SAMPLERATE_CFLAGS",
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[enable_lsr=no;AC_MSG_WARN([libsamplerate not found -- disabling])])
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fi
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if test x$enable_mvp = xyes; then
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AC_DEFINE(HAVE_MVP,1,[Define to enable Hauppauge Media MVP support])
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fi
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@ -777,8 +789,15 @@ if
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fi
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echo ""
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echo " Other features:"
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if test x$enable_lsr = xyes; then
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echo " libsamplerate support .........enabled"
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else
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echo " libsamplerate support .........disabled"
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fi
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if test x$with_zeroconf != xno; then
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echo " Zeroconf support ..............$with_zeroconf"
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else
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@ -126,6 +126,35 @@ This is the gain (in dB) applied to songs with ReplayGain tags.
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.B volume_normalization <yes or no>
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If yes, mpd will normalize the volume of songs as they play. Default is no.
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.TP
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.B samplerate_converter <integer or prefix>
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Specifies the libsamplerate converter to use.
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The supplied value should either be an integer or a prefix of the name of a converter.
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The list of available converters at the time of writing is below.
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More converters may exist, consult the
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documentation of the Secret Rabbit Code libsamplerate (at http://www.mega-nerd.com/SRC/) for details.
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.RS
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.HP
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Best Sinc Interpolator (0)
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Band limited sinc interpolation, best quality, 97dB SNR, 96% BW.
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.HP
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Medium Sinc Interpolator (1)
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Band limited sinc interpolation, medium quality, 97dB SNR, 90% BW.
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.HP
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Fastest Sinc Interpolator (2, default)
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Band limited sinc interpolation, fastest, 97dB SNR, 80% BW.
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.HP
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ZOH Interpolator (3)
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Zero order hold interpolator, very fast, very poor quality with audible distortions.
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.HP
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Linear Interpolator (4)
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Linear interpolator, very fast, poor quality.
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.RE
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.TP
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.B audio_buffer_size <size in KiB>
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This specifies the size of the audio output buffer that mpd uses. The default
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is 2048.
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@ -123,6 +123,12 @@ error_file "~/.mpd/mpd.error"
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#
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#audio_output_format "44100:16:2"
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#
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# Specifies the libsamplerate converter to use (if compiled in),
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# see man 5 mpd.conf for more information.
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#
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#samplerate_converter <integer or prefix>
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#
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################################################################
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@ -157,6 +157,7 @@ void initConf(void)
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registerConfigParam(CONF_REPLAYGAIN, 0, 0);
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registerConfigParam(CONF_REPLAYGAIN_PREAMP, 0, 0);
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registerConfigParam(CONF_VOLUME_NORMALIZATION, 0, 0);
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registerConfigParam(CONF_SAMPLERATE_CONVERTER, 0, 0);
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registerConfigParam(CONF_AUDIO_BUFFER_SIZE, 0, 0);
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registerConfigParam(CONF_BUFFER_BEFORE_PLAY, 0, 0);
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registerConfigParam(CONF_HTTP_BUFFER_SIZE, 0, 0);
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@ -43,6 +43,7 @@
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#define CONF_REPLAYGAIN "replaygain"
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#define CONF_REPLAYGAIN_PREAMP "replaygain_preamp"
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#define CONF_VOLUME_NORMALIZATION "volume_normalization"
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#define CONF_SAMPLERATE_CONVERTER "samplerate_converter"
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#define CONF_AUDIO_BUFFER_SIZE "audio_buffer_size"
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#define CONF_BUFFER_BEFORE_PLAY "buffer_before_play"
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#define CONF_HTTP_BUFFER_SIZE "http_buffer_size"
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109
src/pcm_utils.c
109
src/pcm_utils.c
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@ -21,11 +21,16 @@
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#include "mpd_types.h"
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#include "log.h"
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#include "utils.h"
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#include "conf.h"
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#include <string.h>
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#include <math.h>
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#include <assert.h>
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#ifdef HAVE_LIBSAMPLERATE
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#include <samplerate.h>
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#endif
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void pcm_volumeChange(char *buffer, int bufferSize, AudioFormat * format,
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int volume)
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{
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@ -46,6 +51,9 @@ void pcm_volumeChange(char *buffer, int bufferSize, AudioFormat * format,
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while (bufferSize > 0) {
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temp32 = *buffer16;
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temp32 *= volume;
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temp32 += rand() & 511;
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temp32 -= rand() & 511;
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temp32 += 500;
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temp32 /= 1000;
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*buffer16 = temp32 > 32767 ? 32767 :
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(temp32 < -32768 ? -32768 : temp32);
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@ -57,6 +65,9 @@ void pcm_volumeChange(char *buffer, int bufferSize, AudioFormat * format,
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while (bufferSize > 0) {
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temp32 = *buffer8;
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temp32 *= volume;
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temp32 += rand() & 511;
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temp32 -= rand() & 511;
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temp32 += 500;
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temp32 /= 1000;
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*buffer8 = temp32 > 127 ? 127 :
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(temp32 < -128 ? -128 : temp32);
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@ -86,7 +97,11 @@ static void pcm_add(char *buffer1, char *buffer2, size_t bufferSize1,
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while (bufferSize1 > 0 && bufferSize2 > 0) {
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temp32 =
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(vol1 * (*buffer16_1) +
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vol2 * (*buffer16_2)) / 1000;
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vol2 * (*buffer16_2));
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temp32 += rand() & 511;
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temp32 -= rand() & 511;
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temp32 += 500;
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temp32 /= 1000;
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*buffer16_1 =
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temp32 > 32767 ? 32767 : (temp32 <
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-32768 ? -32768 : temp32);
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@ -101,7 +116,11 @@ static void pcm_add(char *buffer1, char *buffer2, size_t bufferSize1,
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case 8:
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while (bufferSize1 > 0 && bufferSize2 > 0) {
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temp32 =
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(vol1 * (*buffer8_1) + vol2 * (*buffer8_2)) / 1000;
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(vol1 * (*buffer8_1) + vol2 * (*buffer8_2));
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temp32 += rand() & 511;
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temp32 -= rand() & 511;
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temp32 += 500;
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temp32 /= 1000;
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*buffer8_1 =
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temp32 > 127 ? 127 : (temp32 <
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-128 ? -128 : temp32);
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@ -133,6 +152,38 @@ void pcm_mix(char *buffer1, char *buffer2, size_t bufferSize1,
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format);
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}
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#ifdef HAVE_LIBSAMPLERATE
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static int pcm_getSamplerateConverter(void) {
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const char *conf, *test;
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int convalgo = SRC_SINC_FASTEST;
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int newalgo;
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size_t len;
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conf = getConfigParamValue(CONF_SAMPLERATE_CONVERTER);
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if(conf) {
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newalgo = strtol(conf, (char **)&test, 10);
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if(*test) {
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len = strlen(conf);
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for(newalgo = 0; ; newalgo++) {
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test = src_get_name(newalgo);
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if(!test)
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break; /* FAIL */
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if(!strncasecmp(test, conf, len)) {
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convalgo = newalgo;
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break;
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}
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}
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} else {
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if(src_get_name(newalgo))
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convalgo = newalgo;
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/* else FAIL */
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}
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}
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DEBUG("Selecting samplerate converter '%s'\n", src_get_name(convalgo));
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return convalgo;
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}
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#endif
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/* outFormat bits must be 16 and channels must be 2! */
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void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, size_t
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inSize, AudioFormat * outFormat, char *outBuffer)
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@ -234,6 +285,47 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, size_t
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if (inFormat->sampleRate == outFormat->sampleRate) {
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memcpy(outBuffer, dataChannelConv, dataChannelLen);
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} else {
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#ifdef HAVE_LIBSAMPLERATE
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static SRC_STATE *state = NULL;
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static SRC_DATA data;
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int error;
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static double ratio = 0;
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double newratio;
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if(!state) {
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state = src_new(pcm_getSamplerateConverter(), outFormat->channels, &error);
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if(!state) {
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ERROR("Cannot create new samplerate state: %s\n", src_strerror(error));
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exit(EXIT_FAILURE);
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} else {
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DEBUG("Samplerate converter initialized\n");
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}
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}
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newratio = (double)outFormat->sampleRate / (double)inFormat->sampleRate;
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if(newratio != ratio) {
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src_set_ratio(state, ratio = newratio);
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DEBUG("Setting samplerate conversion ratio to %.2lf\n", ratio);
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}
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data.input_frames = dataChannelLen / 2 / outFormat->channels;
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data.output_frames = pcm_sizeOfOutputBufferForAudioFormatConversion(inFormat, dataChannelLen, outFormat) / 2 / outFormat->channels;
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data.src_ratio = (double)data.output_frames / (double)data.input_frames;
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float conversionInBuffer[data.input_frames * outFormat->channels];
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float conversionOutBuffer[data.output_frames * outFormat->channels];
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data.data_in = conversionInBuffer;
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data.data_out = conversionOutBuffer;
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src_short_to_float_array((short *)dataChannelConv, data.data_in, data.input_frames * outFormat->channels);
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error = src_process(state, &data);
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if(error) {
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ERROR("Cannot process samples: %s\n", src_strerror(error));
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exit(EXIT_FAILURE);
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}
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src_float_to_short_array(data.data_out, (short *)outBuffer, data.output_frames * outFormat->channels);
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#else
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/* only works if outFormat is 16-bit stereo! */
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/* resampling code blatantly ripped from ESD */
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mpd_uint32 rd_dat = 0;
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@ -241,11 +333,7 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, size_t
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mpd_sint16 lsample, rsample;
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mpd_sint16 *out = (mpd_sint16 *) outBuffer;
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mpd_sint16 *in = (mpd_sint16 *) dataChannelConv;
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const int shift = sizeof(mpd_sint16) * outFormat->channels;
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mpd_uint32 nlen = (((dataChannelLen / shift) *
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(mpd_uint32) (outFormat->sampleRate)) /
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inFormat->sampleRate);
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nlen *= outFormat->channels;
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mpd_uint32 nlen = pcm_sizeOfOutputBufferForAudioFormatConversion(inFormat, inSize, outFormat) / sizeof(mpd_sint16);
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switch (outFormat->channels) {
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case 1:
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@ -272,6 +360,7 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, size_t
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}
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break;
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}
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#endif
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}
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return;
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@ -306,9 +395,9 @@ size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
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}
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}
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outSize = (((outSize / shift) * (mpd_uint32) (outFormat->sampleRate)) /
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inFormat->sampleRate);
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outSize /= shift;
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outSize = floor(0.5 + (double)outSize *
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((double)outFormat->sampleRate / (double)inFormat->sampleRate));
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outSize *= shift;
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return outSize;
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