/* the Music Player Daemon (MPD) * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com) * This project's homepage is: http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #include "pcm_utils.h" #include "log.h" #include "utils.h" #include "conf.h" #include "audio_format.h" #include <assert.h> #include <string.h> #include <math.h> static inline int pcm_dither(void) { return (rand() & 511) - (rand() & 511); } /** * Check if the value is within the range of the provided bit size, * and caps it if necessary. */ static int32_t pcm_range(int32_t sample, unsigned bits) { if (mpd_unlikely(sample < (-1 << (bits - 1)))) return -1 << (bits - 1); if (mpd_unlikely(sample >= (1 << (bits - 1)))) return (1 << (bits - 1)) - 1; return sample; } static void pcm_volume_change_8(int8_t *buffer, unsigned num_samples, unsigned volume) { while (num_samples > 0) { int32_t sample = *buffer; sample = (sample * volume + pcm_dither() + 500) / 1000; *buffer++ = pcm_range(sample, 8); --num_samples; } } static void pcm_volume_change_16(int16_t *buffer, unsigned num_samples, unsigned volume) { while (num_samples > 0) { int32_t sample = *buffer; sample = (sample * volume + pcm_dither() + 500) / 1000; *buffer++ = pcm_range(sample, 16); --num_samples; } } static void pcm_volume_change_24(int32_t *buffer, unsigned num_samples, unsigned volume) { while (num_samples > 0) { int64_t sample = *buffer; sample = (sample * volume + pcm_dither() + 500) / 1000; *buffer++ = pcm_range(sample, 24); --num_samples; } } void pcm_volumeChange(char *buffer, int bufferSize, const struct audio_format *format, int volume) { if (volume >= 1000) return; if (volume <= 0) { memset(buffer, 0, bufferSize); return; } switch (format->bits) { case 8: pcm_volume_change_8((int8_t *)buffer, bufferSize, volume); break; case 16: pcm_volume_change_16((int16_t *)buffer, bufferSize / 2, volume); break; case 24: pcm_volume_change_24((int32_t*)buffer, bufferSize / 4, volume); break; default: FATAL("%i bits not supported by pcm_volumeChange!\n", format->bits); } } static void pcm_add_8(int8_t *buffer1, const int8_t *buffer2, unsigned num_samples, int volume1, int volume2) { while (num_samples > 0) { int32_t sample1 = *buffer1; int32_t sample2 = *buffer2++; sample1 = ((sample1 * volume1 + sample2 * volume2) + pcm_dither() + 500) / 1000; *buffer1++ = pcm_range(sample1, 8); --num_samples; } } static void pcm_add_16(int16_t *buffer1, const int16_t *buffer2, unsigned num_samples, int volume1, int volume2) { while (num_samples > 0) { int32_t sample1 = *buffer1; int32_t sample2 = *buffer2++; sample1 = ((sample1 * volume1 + sample2 * volume2) + pcm_dither() + 500) / 1000; *buffer1++ = pcm_range(sample1, 16); --num_samples; } } static void pcm_add_24(int32_t *buffer1, const int32_t *buffer2, unsigned num_samples, unsigned volume1, unsigned volume2) { while (num_samples > 0) { int64_t sample1 = *buffer1; int64_t sample2 = *buffer2++; sample1 = ((sample1 * volume1 + sample2 * volume2) + pcm_dither() + 500) / 1000; *buffer1++ = pcm_range(sample1, 24); --num_samples; } } static void pcm_add(char *buffer1, const char *buffer2, size_t size, int vol1, int vol2, const struct audio_format *format) { switch (format->bits) { case 8: pcm_add_8((int8_t *)buffer1, (const int8_t *)buffer2, size, vol1, vol2); break; case 16: pcm_add_16((int16_t *)buffer1, (const int16_t *)buffer2, size / 2, vol1, vol2); break; case 24: pcm_add_24((int32_t*)buffer1, (const int32_t*)buffer2, size / 4, vol1, vol2); break; default: FATAL("%i bits not supported by pcm_add!\n", format->bits); } } void pcm_mix(char *buffer1, const char *buffer2, size_t size, const struct audio_format *format, float portion1) { int vol1; float s = sin(M_PI_2 * portion1); s *= s; vol1 = s * 1000 + 0.5; vol1 = vol1 > 1000 ? 1000 : (vol1 < 0 ? 0 : vol1); pcm_add(buffer1, buffer2, size, vol1, 1000 - vol1, format); } #ifdef HAVE_LIBSAMPLERATE static int pcm_getSampleRateConverter(void) { const char *conf = getConfigParamValue(CONF_SAMPLERATE_CONVERTER); long convalgo; char *test; const char *test2; size_t len; if (!conf) { convalgo = SRC_SINC_FASTEST; goto out; } convalgo = strtol(conf, &test, 10); if (*test == '\0' && src_get_name(convalgo)) goto out; len = strlen(conf); for (convalgo = 0 ; ; convalgo++) { test2 = src_get_name(convalgo); if (!test2) { convalgo = SRC_SINC_FASTEST; break; } if (strncasecmp(test2, conf, len) == 0) goto out; } ERROR("unknown samplerate converter \"%s\"\n", conf); out: DEBUG("selecting samplerate converter \"%s\"\n", src_get_name(convalgo)); return convalgo; } #endif #ifdef HAVE_LIBSAMPLERATE static size_t pcm_convertSampleRate(int8_t channels, uint32_t inSampleRate, const char *inBuffer, size_t inSize, uint32_t outSampleRate, char *outBuffer, size_t outSize, ConvState *convState) { static int convalgo = -1; SRC_DATA *data = &convState->data; size_t dataInSize; size_t dataOutSize; int error; if (convalgo < 0) convalgo = pcm_getSampleRateConverter(); /* (re)set the state/ratio if the in or out format changed */ if ((channels != convState->lastChannels) || (inSampleRate != convState->lastInSampleRate) || (outSampleRate != convState->lastOutSampleRate)) { convState->error = 0; convState->lastChannels = channels; convState->lastInSampleRate = inSampleRate; convState->lastOutSampleRate = outSampleRate; if (convState->state) convState->state = src_delete(convState->state); convState->state = src_new(convalgo, channels, &error); if (!convState->state) { ERROR("cannot create new libsamplerate state: %s\n", src_strerror(error)); convState->error = 1; return 0; } data->src_ratio = (double)outSampleRate / (double)inSampleRate; DEBUG("setting samplerate conversion ratio to %.2lf\n", data->src_ratio); src_set_ratio(convState->state, data->src_ratio); } /* there was an error previously, and nothing has changed */ if (convState->error) return 0; data->input_frames = inSize / 2 / channels; dataInSize = data->input_frames * sizeof(float) * channels; if (dataInSize > convState->dataInSize) { convState->dataInSize = dataInSize; data->data_in = xrealloc(data->data_in, dataInSize); } data->output_frames = outSize / 2 / channels; dataOutSize = data->output_frames * sizeof(float) * channels; if (dataOutSize > convState->dataOutSize) { convState->dataOutSize = dataOutSize; data->data_out = xrealloc(data->data_out, dataOutSize); } src_short_to_float_array((const short *)inBuffer, data->data_in, data->input_frames * channels); error = src_process(convState->state, data); if (error) { ERROR("error processing samples with libsamplerate: %s\n", src_strerror(error)); convState->error = 1; return 0; } src_float_to_short_array(data->data_out, (short *)outBuffer, data->output_frames_gen * channels); return data->output_frames_gen * 2 * channels; } #else /* !HAVE_LIBSAMPLERATE */ /* resampling code blatantly ripped from ESD */ static size_t pcm_convertSampleRate(int8_t channels, uint32_t inSampleRate, const char *inBuffer, mpd_unused size_t inSize, uint32_t outSampleRate, char *outBuffer, size_t outSize, mpd_unused ConvState *convState) { uint32_t rd_dat = 0; uint32_t wr_dat = 0; int16_t *in = (int16_t *)inBuffer; int16_t *out = (int16_t *)outBuffer; uint32_t nlen = outSize / 2; int16_t lsample, rsample; switch (channels) { case 1: while (wr_dat < nlen) { rd_dat = wr_dat * inSampleRate / outSampleRate; lsample = in[rd_dat++]; out[wr_dat++] = lsample; } break; case 2: while (wr_dat < nlen) { rd_dat = wr_dat * inSampleRate / outSampleRate; rd_dat &= ~1; lsample = in[rd_dat++]; rsample = in[rd_dat++]; out[wr_dat++] = lsample; out[wr_dat++] = rsample; } break; } return outSize; } #endif /* !HAVE_LIBSAMPLERATE */ static char *pcm_convertChannels(int8_t channels, const char *inBuffer, size_t inSize, size_t *outSize) { static char *buf; static size_t len; char *outBuffer = NULL; const int16_t *in; int16_t *out; int inSamples, i; switch (channels) { /* convert from 1 -> 2 channels */ case 1: *outSize = (inSize >> 1) << 2; if (*outSize > len) { len = *outSize; buf = xrealloc(buf, len); } outBuffer = buf; inSamples = inSize >> 1; in = (const int16_t *)inBuffer; out = (int16_t *)outBuffer; for (i = 0; i < inSamples; i++) { *out++ = *in; *out++ = *in++; } break; /* convert from 2 -> 1 channels */ case 2: *outSize = inSize >> 1; if (*outSize > len) { len = *outSize; buf = xrealloc(buf, len); } outBuffer = buf; inSamples = inSize >> 2; in = (const int16_t *)inBuffer; out = (int16_t *)outBuffer; for (i = 0; i < inSamples; i++) { *out = (*in++) / 2; *out++ += (*in++) / 2; } break; default: ERROR("only 1 or 2 channels are supported for conversion!\n"); } return outBuffer; } static void pcm_convert_8_to_16(int16_t *out, const int8_t *in, unsigned num_samples) { while (num_samples > 0) { *out++ = *in++ << 8; --num_samples; } } static void pcm_convert_24_to_16(int16_t *out, const int32_t *in, unsigned num_samples) { while (num_samples > 0) { *out++ = *in++ >> 8; --num_samples; } } static const char *pcm_convertTo16bit(int8_t bits, const char *inBuffer, size_t inSize, size_t *outSize) { static char *buf; static size_t len; unsigned num_samples; switch (bits) { case 8: num_samples = inSize; *outSize = inSize << 1; if (*outSize > len) { len = *outSize; buf = xrealloc(buf, len); } pcm_convert_8_to_16((int16_t *)buf, (const int8_t *)inBuffer, num_samples); return buf; case 16: *outSize = inSize; return inBuffer; case 24: num_samples = inSize / 4; *outSize = num_samples * 2; if (*outSize > len) { len = *outSize; buf = xrealloc(buf, len); } pcm_convert_24_to_16((int16_t *)buf, (const int32_t *)inBuffer, num_samples); return buf; } ERROR("only 8 or 16 bits are supported for conversion!\n"); return NULL; } /* outFormat bits must be 16 and channels must be 1 or 2! */ size_t pcm_convertAudioFormat(const struct audio_format *inFormat, const char *inBuffer, size_t inSize, const struct audio_format *outFormat, char *outBuffer, ConvState *convState) { const char *buf; size_t len = 0; size_t outSize = pcm_sizeOfConvBuffer(inFormat, inSize, outFormat); assert(outFormat->bits == 16); assert(outFormat->channels == 2 || outFormat->channels == 1); /* everything else supports 16 bit only, so convert to that first */ buf = pcm_convertTo16bit(inFormat->bits, inBuffer, inSize, &len); if (!buf) exit(EXIT_FAILURE); if (inFormat->channels != outFormat->channels) { buf = pcm_convertChannels(inFormat->channels, buf, len, &len); if (!buf) exit(EXIT_FAILURE); } if (inFormat->sampleRate == outFormat->sampleRate) { assert(outSize >= len); memcpy(outBuffer, buf, len); } else { len = pcm_convertSampleRate(outFormat->channels, inFormat->sampleRate, buf, len, outFormat->sampleRate, outBuffer, outSize, convState); if (len == 0) exit(EXIT_FAILURE); } return len; } size_t pcm_sizeOfConvBuffer(const struct audio_format *inFormat, size_t inSize, const struct audio_format *outFormat) { const double ratio = (double)outFormat->sampleRate / (double)inFormat->sampleRate; const int shift = 2 * outFormat->channels; size_t outSize = inSize; switch (inFormat->bits) { case 8: outSize <<= 1; break; case 16: break; case 24: outSize = (outSize / 4) * 2; break; default: FATAL("only 8 or 16 bits are supported for conversion!\n"); } if (inFormat->channels != outFormat->channels) { switch (inFormat->channels) { case 1: outSize = (outSize >> 1) << 2; break; case 2: outSize >>= 1; break; default: FATAL("only 1 or 2 channels are supported " "for conversion!\n"); } } outSize /= shift; outSize = floor(0.5 + (double)outSize * ratio); outSize *= shift; return outSize; }