diff options
Diffstat (limited to '')
| -rw-r--r-- | src/pcm_utils.c | 109 |
1 files changed, 99 insertions, 10 deletions
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; |