/*************************************************************************** * __________ __ ___. * Open \______ \ ____ ____ | | _\_ |__ _______ ___ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ * \/ \/ \/ \/ \/ * $Id$ * * Copyright (C) 2005 Miika Pekkarinen * * All files in this archive are subject to the GNU General Public License. * See the file COPYING in the source tree root for full license agreement. * * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY * KIND, either express or implied. * ****************************************************************************/ #include #include #include "dsp.h" #include "eq.h" #include "kernel.h" #include "playback.h" #include "system.h" #include "settings.h" #include "replaygain.h" #include "debug.h" /* The "dither" code to convert the 24-bit samples produced by libmad was * taken from the coolplayer project - coolplayer.sourceforge.net */ /* 16-bit samples are scaled based on these constants. The shift should be * no more than 15. */ #define WORD_SHIFT 12 #define WORD_FRACBITS 27 #define NATIVE_DEPTH 16 #define SAMPLE_BUF_SIZE 256 #define RESAMPLE_BUF_SIZE (256 * 4) /* Enough for 11,025 Hz -> 44,100 Hz*/ #define DEFAULT_REPLAYGAIN 0x01000000 /* These are the constants for the filters in the crossfeed */ #define ATT 0x0CCCCCCDL /* 0.1 */ #define ATT_COMP 0x73333333L /* 0.9 */ #define LOW 0x4CCCCCCDL /* 0.6 */ #define LOW_COMP 0x33333333L /* 0.4 */ #define HIGH_NEG -0x66666666L /* -0.2 (not unsigned!) */ #define HIGH_COMP 0x66666666L /* 0.8 */ #if defined(CPU_COLDFIRE) && !defined(SIMULATOR) /* Multiply two S.31 fractional integers and return the sign bit and the * 31 most significant bits of the result. */ #define FRACMUL(x, y) \ ({ \ long t; \ asm volatile ("mac.l %[a], %[b], %%acc0\n\t" \ "movclr.l %%acc0, %[t]\n\t" \ : [t] "=r" (t) : [a] "r" (x), [b] "r" (y)); \ t; \ }) /* Multiply one S.31-bit and one S8.23 fractional integer and return the * sign bit and the 31 most significant bits of the result. */ #define FRACMUL_8(x, y) \ ({ \ long t; \ long u; \ asm volatile ("mac.l %[a], %[b], %%acc0\n\t" \ "move.l %%accext01, %[u]\n\t" \ "movclr.l %%acc0, %[t]\n\t" \ : [t] "=r" (t), [u] "=r" (u) : [a] "r" (x), [b] "r" (y)); \ (t << 8) | (u & 0xff); \ }) /* Multiply one S.31-bit and one S8.23 fractional integer and return the * sign bit and the 31 most significant bits of the result. Load next value * to multiply with into x from s (and increase s); x must contain the * initial value. */ #define FRACMUL_8_LOOP_PART(x, s, d, y) \ { \ long u; \ asm volatile ("mac.l %[a], %[b], (%[c])+, %[a], %%acc0\n\t" \ "move.l %%accext01, %[u]\n\t" \ "movclr.l %%acc0, %[t]" \ : [a] "+r" (x), [c] "+a" (s), [t] "=r" (d), [u] "=r" (u) \ : [b] "r" (y)); \ d = (d << 8) | (u & 0xff); \ } #define FRACMUL_8_LOOP(x, y, s, d) \ { \ long t; \ FRACMUL_8_LOOP_PART(x, s, t, y); \ asm volatile ("move.l %[t],(%[d])+" \ : [d] "+a" (d)\ : [t] "r" (t)); \ } #define ACC(acc, x, y) \ (void)acc; \ asm volatile ("mac.l %[a], %[b], %%acc0" \ : : [a] "i,r" (x), [b] "i,r" (y)); #define GET_ACC(acc) \ ({ \ long t; \ (void)acc; \ asm volatile ("movclr.l %%acc0, %[t]" \ : [t] "=r" (t)); \ t; \ }) #define ACC_INIT(acc, x, y) ACC(acc, x, y) #else #define ACC_INIT(acc, x, y) acc = FRACMUL(x, y) #define ACC(acc, x, y) acc += FRACMUL(x, y) #define GET_ACC(acc) acc #define FRACMUL(x, y) (long) (((((long long) (x)) * ((long long) (y))) >> 31)) #define FRACMUL_8(x, y) (long) (((((long long) (x)) * ((long long) (y))) >> 23)) #define FRACMUL_8_LOOP(x, y, s, d) \ ({ \ long t = x; \ x = *(s)++; \ *(d)++ = (long) (((((long long) (t)) * ((long long) (y))) >> 23)); \ }) #endif struct dsp_config { long codec_frequency; /* Sample rate of data coming from the codec */ long frequency; /* Effective sample rate after pitch shift (if any) */ long clip_min; long clip_max; long track_gain; long album_gain; long track_peak; long album_peak; long replaygain; /* Note that this is in S8.23 format. */ int sample_depth; int sample_bytes; int stereo_mode; int frac_bits; bool dither_enabled; bool new_gain; bool crossfeed_enabled; bool eq_enabled; }; struct resample_data { long phase, delta; long last_sample[2]; }; struct dither_data { long error[3]; long random; }; struct crossfeed_data { long lowpass[2]; long highpass[2]; long delay[2][13]; int index; }; /* Current setup is one lowshelf filters, three peaking filters and one highshelf filter. Varying the number of shelving filters make no sense, but adding peaking filters are possible. */ struct eq_state { char enabled[5]; /* Flags for active filters */ struct eqfilter filters[5]; }; static struct dsp_config dsp_conf[2] IBSS_ATTR; static struct dither_data dither_data[2] IBSS_ATTR; static struct resample_data resample_data[2] IBSS_ATTR; struct crossfeed_data crossfeed_data IBSS_ATTR; static struct eq_state eq_data; static int pitch_ratio = 1000; extern int current_codec; struct dsp_config *dsp; /* The internal format is 32-bit samples, non-interleaved, stereo. This * format is similar to the raw output from several codecs, so the amount * of copying needed is minimized for that case. */ static long sample_buf[SAMPLE_BUF_SIZE] IBSS_ATTR; static long resample_buf[RESAMPLE_BUF_SIZE] IBSS_ATTR; int sound_get_pitch(void) { return pitch_ratio; } void sound_set_pitch(int permille) { pitch_ratio = permille; dsp_configure(DSP_SWITCH_FREQUENCY, (int *)dsp->codec_frequency); } /* Convert at most count samples to the internal format, if needed. Returns * number of samples ready for further processing. Updates src to point * past the samples "consumed" and dst is set to point to the samples to * consume. Note that for mono, dst[0] equals dst[1], as there is no point * in processing the same data twice. */ static int convert_to_internal(const char* src[], int count, long* dst[]) { count = MIN(SAMPLE_BUF_SIZE / 2, count); if ((dsp->sample_depth <= NATIVE_DEPTH) || (dsp->stereo_mode == STEREO_INTERLEAVED)) { dst[0] = &sample_buf[0]; dst[1] = (dsp->stereo_mode == STEREO_MONO) ? dst[0] : &sample_buf[SAMPLE_BUF_SIZE / 2]; } else { dst[0] = (long*) src[0]; dst[1] = (long*) ((dsp->stereo_mode == STEREO_MONO) ? src[0] : src[1]); } if (dsp->sample_depth <= NATIVE_DEPTH) { short* s0 = (short*) src[0]; long* d0 = dst[0]; long* d1 = dst[1]; int scale = WORD_SHIFT; int i; if (dsp->stereo_mode == STEREO_INTERLEAVED) { for (i = 0; i < count; i++) { *d0++ = *s0++ << scale; *d1++ = *s0++ << scale; } } else if (dsp->stereo_mode == STEREO_NONINTERLEAVED) { short* s1 = (short*) src[1]; for (i = 0; i < count; i++) { *d0++ = *s0++ << scale; *d1++ = *s1++ << scale; } } else { for (i = 0; i < count; i++) { *d0++ = *s0++ << scale; } } } else if (dsp->stereo_mode == STEREO_INTERLEAVED) { long* s0 = (long*) src[0]; long* d0 = dst[0]; long* d1 = dst[1]; int i; for (i = 0; i < count; i++) { *d0++ = *s0++; *d1++ = *s0++; } } if (dsp->stereo_mode == STEREO_NONINTERLEAVED) { src[0] += count * dsp->sample_bytes; src[1] += count * dsp->sample_bytes; } else if (dsp->stereo_mode == STEREO_INTERLEAVED) { src[0] += count * dsp->sample_bytes * 2; } else { src[0] += count * dsp->sample_bytes; } return count; } static void resampler_set_delta(int frequency) { resample_data[current_codec].delta = (unsigned long) frequency * 65536LL / NATIVE_FREQUENCY; } /* Linear resampling that introduces a one sample delay, because of our * inability to look into the future at the end of a frame. */ /* TODO: we really should have a separate set of resample functions for both mono and stereo to avoid all this internal branching and looping. */ static long downsample(long **dst, long **src, int count, struct resample_data *r) { long phase = r->phase; long delta = r->delta; long last_sample; long *d[2] = { dst[0], dst[1] }; int pos = phase >> 16; int i = 1, j; int num_channels = dsp->stereo_mode == STEREO_MONO ? 1 : 2; for (j = 0; j < num_channels; j++) { last_sample = r->last_sample[j]; /* Do we need last sample of previous frame for interpolation? */ if (pos > 0) { last_sample = src[j][pos - 1]; } *d[j]++ = last_sample + FRACMUL((phase & 0xffff) << 15, src[j][pos] - last_sample); } phase += delta; while ((pos = phase >> 16) < count) { for (j = 0; j < num_channels; j++) *d[j]++ = src[j][pos - 1] + FRACMUL((phase & 0xffff) << 15, src[j][pos] - src[j][pos - 1]); phase += delta; i++; } /* Wrap phase accumulator back to start of next frame. */ r->phase = phase - (count << 16); r->delta = delta; r->last_sample[0] = src[0][count - 1]; r->last_sample[1] = src[1][count - 1]; return i; } static long upsample(long **dst, long **src, int count, struct resample_data *r) { long phase = r->phase; long delta = r->delta; long *d[2] = { dst[0], dst[1] }; int i = 0, j; int pos; int num_channels = dsp->stereo_mode == STEREO_MONO ? 1 : 2; while ((pos = phase >> 16) == 0) { for (j = 0; j < num_channels; j++) *d[j]++ = r->last_sample[j] + FRACMUL((phase & 0xffff) << 15, src[j][pos] - r->last_sample[j]); phase += delta; i++; } while ((pos = phase >> 16) < count) { for (j = 0; j < num_channels; j++) *d[j]++ = src[j][pos - 1] + FRACMUL((phase & 0xffff) << 15, src[j][pos] - src[j][pos - 1]); phase += delta; i++; } /* Wrap phase accumulator back to start of next frame. */ r->phase = phase - (count << 16); r->delta = delta; r->last_sample[0] = src[0][count - 1]; r->last_sample[1] = src[1][count - 1]; return i; } /* Resample count stereo samples. Updates the src array, if resampling is * done, to refer to the resampled data. Returns number of stereo samples * for further processing. */ static inline int resample(long* src[], int count) { long new_count; if (dsp->frequency != NATIVE_FREQUENCY) { long* dst[2] = {&resample_buf[0], &resample_buf[RESAMPLE_BUF_SIZE / 2]}; if (dsp->frequency < NATIVE_FREQUENCY) { new_count = upsample(dst, src, count, &resample_data[current_codec]); } else { new_count = downsample(dst, src, count, &resample_data[current_codec]); } src[0] = dst[0]; if (dsp->stereo_mode != STEREO_MONO) src[1] = dst[1]; else src[1] = dst[0]; } else { new_count = count; } return new_count; } static inline long clip_sample(long sample, long min, long max) { if (sample > max) { sample = max; } else if (sample < min) { sample = min; } return sample; } /* The "dither" code to convert the 24-bit samples produced by libmad was * taken from the coolplayer project - coolplayer.sourceforge.net */ static long dither_sample(long sample, long bias, long mask, struct dither_data* dither) { long output; long random; long min; long max; /* Noise shape and bias */ sample += dither->error[0] - dither->error[1] + dither->error[2]; dither->error[2] = dither->error[1]; dither->error[1] = dither->error[0] / 2; output = sample + bias; /* Dither */ random = dither->random * 0x0019660dL + 0x3c6ef35fL; sample += (random & mask) - (dither->random & mask); dither->random = random; /* Clip and quantize */ min = dsp->clip_min; max = dsp->clip_max; sample = clip_sample(sample, min, max); output = clip_sample(output, min, max) & ~mask; /* Error feedback */ dither->error[0] = sample - output; return output; } /* Apply a constant gain to the samples (e.g., for ReplayGain). May update * the src array if gain was applied. * Note that this must be called before the resampler. */ #if defined(CPU_COLDFIRE) && !defined(SIMULATOR) static const long crossfeed_coefs[6] ICONST_ATTR = { LOW, LOW_COMP, HIGH_NEG, HIGH_COMP, ATT, ATT_COMP }; static void apply_crossfeed(long* src[], int count) { asm volatile ( "lea.l crossfeed_data, %%a1 \n" "lea.l (16, %%a1), %%a0 \n" "movem.l (%%a1), %%d0-%%d3 \n" "move.l (120, %%a1), %%d4 \n" /* fetch left, right, LOW and LOW_COMP for first iteration */ "move.l (%[src0]), %%d5 \n" "move.l (%[src1]), %%d6 \n" "move.l (%[coef])+, %%a1 \n" "move.l (%[coef])+, %%a2 \n" /* Register usage in loop: * a0 = &delay[0][0], a1 & a2 = coefs * d0 = low_left, d1 = low_right, * d2 = high_left, d3 = high_right, * d4 = delay line index, * d5 = src[0][i], d6 = src[1][i]. * The rest are described in asm constraint list. */ ".cfloop:" /* LOW*low_left + LOW_COMP*left */ "mac.l %%a1, %%d0, %%acc0 \n" "mac.l %%a2, %%d5, %%acc0 \n" /* LOW*low_right + LOW_COMP*right */ "mac.l %%a1, %%d1, (%[coef])+, %%a1, %%acc1 \n" /* a1 = HIGH_NEG */ "mac.l %%a2, %%d6, (%[coef])+, %%a2, %%acc1 \n" /* a2 = HIGH_COMP */ "movclr.l %%acc0, %%d0 \n" /* get low_left */ "movclr.l %%acc1, %%d1 \n" /* get low_right */ /* HIGH_NEG*high_left + HIGH_COMP*left */ "mac.l %%a1, %%d2, %%acc0 \n" "mac.l %%a2, %%d5, %%acc0 \n" /* HIGH_NEG*high_right + HIGH_COMP*right */ "mac.l %%a1, %%d3, (%[coef])+, %%a1, %%acc1 \n" /* a1 = ATT */ "mac.l %%a2, %%d6, (%[coef])+, %%a2, %%acc1 \n" /* a2 = ATT_COMP */ "lea.l (-6*4, %[coef]), %[coef] \n" /* coef = &coefs[0] */ "move.l (%%a0, %%d4*4), %%a3 \n" /* a3=delay[0][idx] */ "move.l (52, %%a0, %%d4*4), %%d5 \n" /* d5=delay[1][idx] */ "movclr.l %%acc0, %%d2 \n" /* get high_left */ "movclr.l %%acc1, %%d3 \n" /* get high_right */ /* ATT*delay_r + ATT_COMP*high_left */ "mac.l %%a1, %%d5, (4, %[src0]), %%d5, %%acc0\n" /* d5 = src[0][i+1] */ "mac.l %%a2, %%d2, (4, %[src1]), %%d6, %%acc0\n" /* d6 = src[1][i+1] */ /* ATT*delay_l + ATT_COMP*high_right */ "mac.l %%a1, %%a3, (%[coef])+, %%a1, %%acc1 \n" /* a1 = LOW */ "mac.l %%a2, %%d3, (%[coef])+, %%a2, %%acc1 \n" /* a2 = LOW_COMP */ /* save crossfed samples to output */ "movclr.l %%acc0, %%a3 \n" "move.l %%a3, (%[src0])+ \n" /* src[0][i++] = out_l */ "movclr.l %%acc1, %%a3 \n" "move.l %%a3, (%[src1])+ \n" /* src[1][i++] = out_r */ "move.l %%d0, (%%a0, %%d4*4) \n" /* delay[0][index] = low_left */ "move.l %%d1, (52, %%a0, %%d4*4)\n" /* delay[1][index] = low_right */ "addq.l #1, %%d4 \n" /* index++ */ "cmp.l #13, %%d4 \n" /* if (index >= 13) { */ "jlt .nowrap \n" "clr.l %%d4 \n" /* index = 0 */ ".nowrap: \n" /* } */ "subq.l #1, %[count] \n" "jne .cfloop \n" /* save data back to struct */ "lea.l crossfeed_data, %%a1 \n" "movem.l %%d0-%%d3, (%%a1) \n" "move.l %%d4, (120, %%a1) \n" /* NOTE: We _just_ have enough registers for our use here, clobber just one more and GCC will fail. */ : : [count] "d" (count), [src0] "a" (src[0]), [src1] "a" (src[1]), [coef] "a" (crossfeed_coefs) : "d0", "d1", "d2", "d3", "d4", "d5", "d6", "a0", "a1", "a2", "a3" ); } #else static void apply_crossfeed(long* src[], int count) { long a; /* accumulator */ long low_left = crossfeed_data.lowpass[0]; long low_right = crossfeed_data.lowpass[1]; long high_left = crossfeed_data.highpass[0]; long high_right = crossfeed_data.highpass[1]; unsigned int index = crossfeed_data.index; long left, right; long * delay_l = crossfeed_data.delay[0]; long * delay_r = crossfeed_data.delay[1]; int i; for (i = 0; i < count; i++) { /* use a low-pass filter on the signal */ left = src[0][i]; right = src[1][i]; ACC_INIT(a, LOW, low_left); ACC(a, LOW_COMP, left); low_left = GET_ACC(a); ACC_INIT(a, LOW, low_right); ACC(a, LOW_COMP, right); low_right = GET_ACC(a); /* use a high-pass filter on the signal */ ACC_INIT(a, HIGH_NEG, high_left); ACC(a, HIGH_COMP, left); high_left = GET_ACC(a); ACC_INIT(a, HIGH_NEG, high_right); ACC(a, HIGH_COMP, right); high_right = GET_ACC(a); /* New data is the high-passed signal + delayed and attenuated * low-passed signal from the other channel */ ACC_INIT(a, ATT, delay_r[index]); ACC(a, ATT_COMP, high_left); src[0][i] = GET_ACC(a); ACC_INIT(a, ATT, delay_l[index]); ACC(a, ATT_COMP, high_right); src[1][i] = GET_ACC(a); /* Store the low-passed signal in the ringbuffer */ delay_l[index] = low_left; delay_r[index] = low_right; index = (index + 1) % 13; } crossfeed_data.index = index; crossfeed_data.lowpass[0] = low_left; crossfeed_data.lowpass[1] = low_right; crossfeed_data.highpass[0] = high_left; crossfeed_data.highpass[1] = high_right; } #endif #define EQ_CUTOFF_USER2REAL(x) (0xffffffff / NATIVE_FREQUENCY * (x)) #define EQ_Q_USER2REAL(x) (((x) << 16) / 10) #define EQ_GAIN_USER2REAL(x) (((x) << 16) / 10) /* Synchronize the EQ filters with the global settings */ void dsp_eq_update_data(bool enabled, int band) { int *setting; int gain, cutoff, q; dsp->eq_enabled = enabled; /* Adjust setting pointer to the band we actually want to change */ setting = &global_settings.eq_band0_cutoff + (band * 3); cutoff = *setting++; q = *setting++; gain = *setting++; DEBUGF("cutoff %d, q %d, gain %d\n", cutoff, q, gain); #if defined(CPU_COLDFIRE) && !defined(SIMULATOR) /* set emac unit for dsp processing, and save old macsr, we're running in codec thread context at this point, so can't clobber it */ unsigned long old_macsr = coldfire_get_macsr(); coldfire_set_macsr(EMAC_FRACTIONAL | EMAC_SATURATE | EMAC_ROUND); #endif if (gain == 0) { eq_data.enabled[band] = 0; } else { if (band == 0) eq_ls_coefs(EQ_CUTOFF_USER2REAL(cutoff), EQ_Q_USER2REAL(q), EQ_GAIN_USER2REAL(gain), eq_data.filters[band].coefs); else if (band == 4) eq_hs_coefs(EQ_CUTOFF_USER2REAL(cutoff), EQ_Q_USER2REAL(q), EQ_GAIN_USER2REAL(gain), eq_data.filters[band].coefs); else eq_pk_coefs(EQ_CUTOFF_USER2REAL(cutoff), EQ_Q_USER2REAL(q), EQ_GAIN_USER2REAL(gain), eq_data.filters[band].coefs); eq_data.enabled[band] = 1; } #if defined(CPU_COLDFIRE) && !defined(SIMULATOR) /* set old macsr again */ coldfire_set_macsr(old_macsr); #endif } /* Apply EQ filters to those bands that have got it switched on. */ static void eq_process(long **x, unsigned num) { int i; unsigned int channels = dsp->stereo_mode != STEREO_MONO ? 2 : 1; unsigned shift; /* filter configuration currently is 1 low shelf filter, 3 band peaking filters and 1 high shelf filter, in that order. we need to know this so we can choose the correct shift factor. */ for (i = 0; i < 5; i++) { if (eq_data.enabled[i]) { if (i == 0 || i == 4) /* shelving filters */ shift = EQ_SHELF_SHIFT; else shift = EQ_PEAK_SHIFT; eq_filter(x, &eq_data.filters[i], num, channels, shift); } } } /* Apply a constant gain to the samples (e.g., for ReplayGain). May update * the src array if gain was applied. * Note that this must be called before the resampler. */ static void apply_gain(long* _src[], int _count) { struct dsp_config *my_dsp = dsp; if (my_dsp->replaygain) { long** src = _src; int count = _count; long* s0 = src[0]; long* s1 = src[1]; long gain = my_dsp->replaygain; long s; int i; long *d; if (s0 != s1) { d = &sample_buf[SAMPLE_BUF_SIZE / 2]; src[1] = d; s = *s1++; for (i = 0; i < count; i++) FRACMUL_8_LOOP(s, gain, s1, d); } else { src[1] = &sample_buf[0]; } d = &sample_buf[0]; src[0] = d; s = *s0++; for (i = 0; i < count; i++) FRACMUL_8_LOOP(s, gain, s0, d); } } static void write_samples(short* dst, long* src[], int count) { long* s0 = src[0]; long* s1 = src[1]; int scale = dsp->frac_bits + 1 - NATIVE_DEPTH; if (dsp->dither_enabled) { long bias = (1L << (dsp->frac_bits - NATIVE_DEPTH)); long mask = (1L << scale) - 1; while (count-- > 0) { *dst++ = (short) (dither_sample(*s0++, bias, mask, &dither_data[0]) >> scale); *dst++ = (short) (dither_sample(*s1++, bias, mask, &dither_data[1]) >> scale); } } else { long min = dsp->clip_min; long max = dsp->clip_max; while (count-- > 0) { *dst++ = (short) (clip_sample(*s0++, min, max) >> scale); *dst++ = (short) (clip_sample(*s1++, min, max) >> scale); } } } /* Process and convert src audio to dst based on the DSP configuration, * reading size bytes of audio data. dst is assumed to be large enough; use * dst_get_dest_size() to get the required size. src is an array of * pointers; for mono and interleaved stereo, it contains one pointer to the * start of the audio data; for non-interleaved stereo, it contains two * pointers, one for each audio channel. Returns number of bytes written to * dest. */ long dsp_process(char* dst, const char* src[], long size) { long* tmp[2]; long written = 0; long factor; int samples; #if defined(CPU_COLDFIRE) && !defined(SIMULATOR) /* set emac unit for dsp processing, and save old macsr, we're running in codec thread context at this point, so can't clobber it */ unsigned long old_macsr = coldfire_get_macsr(); coldfire_set_macsr(EMAC_FRACTIONAL | EMAC_SATURATE); #endif dsp = &dsp_conf[current_codec]; factor = (dsp->stereo_mode != STEREO_MONO) ? 2 : 1; size /= dsp->sample_bytes * factor; dsp_set_replaygain(false); while (size > 0) { samples = convert_to_internal(src, size, tmp); size -= samples; apply_gain(tmp, samples); samples = resample(tmp, samples); if (dsp->crossfeed_enabled && dsp->stereo_mode != STEREO_MONO) apply_crossfeed(tmp, samples); if (dsp->eq_enabled) eq_process(tmp, samples); write_samples((short*) dst, tmp, samples); written += samples; dst += samples * sizeof(short) * 2; yield(); } #if defined(CPU_COLDFIRE) && !defined(SIMULATOR) /* set old macsr again */ coldfire_set_macsr(old_macsr); #endif return written * sizeof(short) * 2; } /* Given size bytes of input data, calculate the maximum number of bytes of * output data that would be generated (the calculation is not entirely * exact and rounds upwards to be on the safe side; during resampling, * the number of samples generated depends on the current state of the * resampler). */ /* dsp_input_size MUST be called afterwards */ long dsp_output_size(long size) { dsp = &dsp_conf[current_codec]; if (dsp->sample_depth > NATIVE_DEPTH) { size /= 2; } if (dsp->frequency != NATIVE_FREQUENCY) { size = (long) ((((unsigned long) size * NATIVE_FREQUENCY) + (dsp->frequency - 1)) / dsp->frequency); } /* round to the next multiple of 2 (these are shorts) */ size = (size + 1) & ~1; if (dsp->stereo_mode == STEREO_MONO) { size *= 2; } /* now we have the size in bytes for two resampled channels, * and the size in (short) must not exceed RESAMPLE_BUF_SIZE to * avoid resample buffer overflow. One must call dsp_input_size() * to get the correct input buffer size. */ if (size > RESAMPLE_BUF_SIZE*2) size = RESAMPLE_BUF_SIZE*2; return size; } /* Given size bytes of output buffer, calculate number of bytes of input * data that would be consumed in order to fill the output buffer. */ long dsp_input_size(long size) { dsp = &dsp_conf[current_codec]; /* convert to number of output stereo samples. */ size /= 2; /* Mono means we need half input samples to fill the output buffer */ if (dsp->stereo_mode == STEREO_MONO) size /= 2; /* size is now the number of resampled input samples. Convert to original input samples. */ if (dsp->frequency != NATIVE_FREQUENCY) { /* Use the real resampling delta = * (unsigned long) dsp->frequency * 65536 / NATIVE_FREQUENCY, and * round towards zero to avoid buffer overflows. */ size = ((unsigned long)size * resample_data[current_codec].delta) >> 16; } /* Convert back to bytes. */ if (dsp->sample_depth > NATIVE_DEPTH) size *= 4; else size *= 2; return size; } int dsp_stereo_mode(void) { dsp = &dsp_conf[current_codec]; return dsp->stereo_mode; } bool dsp_configure(int setting, void *value) { dsp = &dsp_conf[current_codec]; switch (setting) { case DSP_SET_FREQUENCY: memset(&resample_data[current_codec], 0, sizeof(struct resample_data)); /* Fall through!!! */ case DSP_SWITCH_FREQUENCY: dsp->codec_frequency = ((int) value == 0) ? NATIVE_FREQUENCY : (int) value; /* Account for playback speed adjustment when settingg dsp->frequency if we're called from the main audio thread. Voice UI thread should not need this feature. */ if (current_codec == CODEC_IDX_AUDIO) dsp->frequency = pitch_ratio * dsp->codec_frequency / 1000; else dsp->frequency = dsp->codec_frequency; resampler_set_delta(dsp->frequency); break; case DSP_SET_CLIP_MIN: dsp->clip_min = (long) value; break; case DSP_SET_CLIP_MAX: dsp->clip_max = (long) value; break; case DSP_SET_SAMPLE_DEPTH: dsp->sample_depth = (long) value; if (dsp->sample_depth <= NATIVE_DEPTH) { dsp->frac_bits = WORD_FRACBITS; dsp->sample_bytes = sizeof(short); dsp->clip_max = ((1 << WORD_FRACBITS) - 1); dsp->clip_min = -((1 << WORD_FRACBITS)); } else { dsp->frac_bits = (long) value; dsp->sample_bytes = sizeof(long); dsp->clip_max = (1 << (long)value) - 1; dsp->clip_min = -(1 << (long)value); } break; case DSP_SET_STEREO_MODE: dsp->stereo_mode = (long) value; break; case DSP_RESET: dsp->dither_enabled = false; dsp->stereo_mode = STEREO_NONINTERLEAVED; dsp->clip_max = ((1 << WORD_FRACBITS) - 1); dsp->clip_min = -((1 << WORD_FRACBITS)); dsp->track_gain = 0; dsp->album_gain = 0; dsp->track_peak = 0; dsp->album_peak = 0; dsp->codec_frequency = dsp->frequency = NATIVE_FREQUENCY; dsp->sample_depth = NATIVE_DEPTH; dsp->frac_bits = WORD_FRACBITS; dsp->new_gain = true; break; case DSP_DITHER: memset(dither_data, 0, sizeof(dither_data)); dsp->dither_enabled = (bool) value; break; case DSP_SET_TRACK_GAIN: dsp->track_gain = (long) value; dsp->new_gain = true; break; case DSP_SET_ALBUM_GAIN: dsp->album_gain = (long) value; dsp->new_gain = true; break; case DSP_SET_TRACK_PEAK: dsp->track_peak = (long) value; dsp->new_gain = true; break; case DSP_SET_ALBUM_PEAK: dsp->album_peak = (long) value; dsp->new_gain = true; break; default: return 0; } return 1; } void dsp_set_crossfeed(bool enable) { if (enable) memset(&crossfeed_data, 0, sizeof(crossfeed_data)); dsp->crossfeed_enabled = enable; } void dsp_set_replaygain(bool always) { dsp = &dsp_conf[current_codec]; if (always || dsp->new_gain) { long gain = 0; dsp->new_gain = false; if (global_settings.replaygain || global_settings.replaygain_noclip) { bool track_mode = ((global_settings.replaygain_type == REPLAYGAIN_TRACK) || ((global_settings.replaygain_type == REPLAYGAIN_SHUFFLE) && global_settings.playlist_shuffle)); long peak = (track_mode || !dsp->album_peak) ? dsp->track_peak : dsp->album_peak; if (global_settings.replaygain) { gain = (track_mode || !dsp->album_gain) ? dsp->track_gain : dsp->album_gain; if (global_settings.replaygain_preamp) { long preamp = get_replaygain_int( global_settings.replaygain_preamp * 10); gain = (long) (((int64_t) gain * preamp) >> 24); } } if (gain == 0) { /* So that noclip can work even with no gain information. */ gain = DEFAULT_REPLAYGAIN; } if (global_settings.replaygain_noclip && (peak != 0) && ((((int64_t) gain * peak) >> 24) >= DEFAULT_REPLAYGAIN)) { gain = (((int64_t) DEFAULT_REPLAYGAIN << 24) / peak); } if (gain == DEFAULT_REPLAYGAIN) { /* Nothing to do, disable processing. */ gain = 0; } } /* Store in S8.23 format to simplify calculations. */ dsp->replaygain = gain >> 1; } }