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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.

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git-svn-id: https://svn.musicpd.org/mpd/trunk@5316 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This commit is contained in:
Avuton Olrich
2007-02-02 03:51:07 +00:00
parent 96c5976ccc
commit 79ef8ba248
6 changed files with 156 additions and 11 deletions
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1
src/conf.c
View File

@@ -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);

1
src/conf.h
View File

@@ -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"

109
src/pcm_utils.c
View File

@@ -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;