/*************************************************************************** * __________ __ ___. * 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 "config.h" #include #include #include #include #include "dsp.h" #include "eq.h" #include "kernel.h" #include "playback.h" #include "system.h" #include "settings.h" #include "replaygain.h" #include "misc.h" #include "debug.h" /* 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 /* If the buffer sizes change, check the assembly code! */ #define SAMPLE_BUF_COUNT 256 #define RESAMPLE_BUF_COUNT (256 * 4) /* Enough for 11,025 Hz -> 44,100 Hz*/ #define DEFAULT_GAIN 0x01000000 #define SAMPLE_BUF_LEFT_CHANNEL 0 #define SAMPLE_BUF_RIGHT_CHANNEL (SAMPLE_BUF_COUNT/2) #define RESAMPLE_BUF_LEFT_CHANNEL 0 #define RESAMPLE_BUF_RIGHT_CHANNEL (RESAMPLE_BUF_COUNT/2) /* enums to index conversion properly with stereo mode and other settings */ enum { SAMPLE_INPUT_LE_NATIVE_I_STEREO = STEREO_INTERLEAVED, SAMPLE_INPUT_LE_NATIVE_NI_STEREO = STEREO_NONINTERLEAVED, SAMPLE_INPUT_LE_NATIVE_MONO = STEREO_MONO, SAMPLE_INPUT_GT_NATIVE_I_STEREO = STEREO_INTERLEAVED + STEREO_NUM_MODES, SAMPLE_INPUT_GT_NATIVE_NI_STEREO = STEREO_NONINTERLEAVED + STEREO_NUM_MODES, SAMPLE_INPUT_GT_NATIVE_MONO = STEREO_MONO + STEREO_NUM_MODES, SAMPLE_INPUT_GT_NATIVE_1ST_INDEX = STEREO_NUM_MODES }; enum { SAMPLE_OUTPUT_MONO = 0, SAMPLE_OUTPUT_STEREO, SAMPLE_OUTPUT_DITHERED_MONO, SAMPLE_OUTPUT_DITHERED_STEREO }; /**************************************************************************** * NOTE: Any assembly routines that use these structures must be updated * if current data members are moved or changed. */ struct resample_data { uint32_t delta; /* 00h */ uint32_t phase; /* 04h */ int32_t last_sample[2]; /* 08h */ /* 10h */ }; /* This is for passing needed data to assembly dsp routines. If another * dsp parameter needs to be passed, add to the end of the structure * and remove from dsp_config. * If another function type becomes assembly optimized and requires dsp * config info, add a pointer paramter of type "struct dsp_data *". * If removing something from other than the end, reserve the spot or * else update every implementation for every target. * Be sure to add the offset of the new member for easy viewing as well. :) * It is the first member of dsp_config and all members can be accessesed * through the main aggregate but this is intended to make a safe haven * for these items whereas the c part can be rearranged at will. dsp_data * could even moved within dsp_config without disurbing the order. */ struct dsp_data { int output_scale; /* 00h */ int num_channels; /* 04h */ struct resample_data resample_data; /* 08h */ int32_t clip_min; /* 18h */ int32_t clip_max; /* 1ch */ int32_t gain; /* 20h - Note that this is in S8.23 format. */ /* 24h */ }; /* No asm...yet */ struct dither_data { long error[3]; /* 00h */ long random; /* 0ch */ /* 10h */ }; struct crossfeed_data { int32_t gain; /* 00h - Direct path gain */ int32_t coefs[3]; /* 04h - Coefficients for the shelving filter */ int32_t history[4]; /* 10h - Format is x[n - 1], y[n - 1] for both channels */ int32_t delay[13][2]; /* 20h */ int32_t *index; /* 88h - Current pointer into the delay line */ /* 8ch */ }; /* 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 is possible. */ struct eq_state { char enabled[5]; /* 00h - Flags for active filters */ struct eqfilter filters[5]; /* 08h - packing is 4? */ /* 10ch */ }; /* Include header with defines which functions are implemented in assembly code for the target */ #include /* Typedefs keep things much neater in this case */ typedef void (*sample_input_fn_type)(int count, const char *src[], int32_t *dst[]); typedef int (*resample_fn_type)(int count, struct dsp_data *data, int32_t *src[], int32_t *dst[]); typedef void (*sample_output_fn_type)(int count, struct dsp_data *data, int32_t *src[], int16_t *dst); /* Single-DSP channel processing in place */ typedef void (*channels_process_fn_type)(int count, int32_t *buf[]); /* DSP local channel processing in place */ typedef void (*channels_process_dsp_fn_type)(int count, struct dsp_data *data, int32_t *buf[]); /* ***************************************************************************/ struct dsp_config { struct dsp_data data; /* Config members for use in asm routines */ long codec_frequency; /* Sample rate of data coming from the codec */ long frequency; /* Effective sample rate after pitch shift (if any) */ int sample_depth; int sample_bytes; int stereo_mode; int frac_bits; /* Functions that change depending upon settings - NULL if stage is disabled */ sample_input_fn_type input_samples; resample_fn_type resample; sample_output_fn_type output_samples; /* These will be NULL for the voice codec and is more economical that way */ channels_process_dsp_fn_type apply_gain; channels_process_fn_type apply_crossfeed; channels_process_fn_type channels_process; }; /* General DSP config */ static struct dsp_config dsp_conf[2] IBSS_ATTR; /* 0=A, 1=V */ /* Dithering */ static struct dither_data dither_data[2] IBSS_ATTR; /* 0=left, 1=right */ static long dither_mask IBSS_ATTR; static long dither_bias IBSS_ATTR; /* Crossfeed */ struct crossfeed_data crossfeed_data IDATA_ATTR = /* A */ { .index = (int32_t *)crossfeed_data.delay }; /* Equalizer */ static struct eq_state eq_data; /* A/V */ #ifdef HAVE_SW_TONE_CONTROLS static int prescale; static int bass; static int treble; /* Filter struct for software bass/treble controls */ static struct eqfilter tone_filter; #endif /* Settings applicable to audio codec only */ static int pitch_ratio = 1000; static int channels_mode; long dsp_sw_gain; long dsp_sw_cross; static bool dither_enabled; static bool eq_enabled IBSS_ATTR; static long eq_precut; static long track_gain; static bool new_gain; static long album_gain; static long track_peak; static long album_peak; static long replaygain; static bool crossfeed_enabled; #define audio_dsp (&dsp_conf[CODEC_IDX_AUDIO]) #define voice_dsp (&dsp_conf[CODEC_IDX_VOICE]) static struct dsp_config *dsp IDATA_ATTR = audio_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. */ int32_t sample_buf[SAMPLE_BUF_COUNT] IBSS_ATTR; static int32_t resample_buf[RESAMPLE_BUF_COUNT] IBSS_ATTR; /* set a new dsp and return old one */ static inline struct dsp_config * switch_dsp(struct dsp_config *_dsp) { struct dsp_config * old_dsp = dsp; dsp = _dsp; return old_dsp; } #if 0 /* Clip sample to arbitrary limits where range > 0 and min + range = max */ static inline long clip_sample(int32_t sample, int32_t min, int32_t range) { int32_t c = sample - min; if ((uint32_t)c > (uint32_t)range) { sample -= c; if (c > 0) sample += range; } return sample; } #endif /* Clip sample to signed 16 bit range */ static inline int32_t clip_sample_16(int32_t sample) { if ((int16_t)sample != sample) sample = 0x7fff ^ (sample >> 31); return sample; } int sound_get_pitch(void) { return pitch_ratio; } void sound_set_pitch(int permille) { pitch_ratio = permille; dsp_configure(DSP_SWITCH_FREQUENCY, dsp->codec_frequency); } /* Convert count samples to the internal format, if needed. 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. */ /* convert count 16-bit mono to 32-bit mono */ static void sample_input_lte_native_mono( int count, const char *src[], int32_t *dst[]) { const int16_t *s = (int16_t *) src[0]; const int16_t * const send = s + count; int32_t *d = dst[0] = dst[1] = &sample_buf[SAMPLE_BUF_LEFT_CHANNEL]; int scale = WORD_SHIFT; do { *d++ = *s++ << scale; } while (s < send); src[0] = (char *)s; } /* convert count 16-bit interleaved stereo to 32-bit noninterleaved */ static void sample_input_lte_native_i_stereo( int count, const char *src[], int32_t *dst[]) { const int32_t *s = (int32_t *) src[0]; const int32_t * const send = s + count; int32_t *dl = dst[0] = &sample_buf[SAMPLE_BUF_LEFT_CHANNEL]; int32_t *dr = dst[1] = &sample_buf[SAMPLE_BUF_RIGHT_CHANNEL]; int scale = WORD_SHIFT; do { int32_t slr = *s++; #ifdef ROCKBOX_LITTLE_ENDIAN *dl++ = (slr >> 16) << scale; *dr++ = (int32_t)(int16_t)slr << scale; #else /* ROCKBOX_BIG_ENDIAN */ *dl++ = (int32_t)(int16_t)slr << scale; *dr++ = (slr >> 16) << scale; #endif } while (s < send); src[0] = (char *)s; } /* convert count 16-bit noninterleaved stereo to 32-bit noninterleaved */ static void sample_input_lte_native_ni_stereo( int count, const char *src[], int32_t *dst[]) { const int16_t *sl = (int16_t *) src[0]; const int16_t *sr = (int16_t *) src[1]; const int16_t * const slend = sl + count; int32_t *dl = dst[0] = &sample_buf[SAMPLE_BUF_LEFT_CHANNEL]; int32_t *dr = dst[1] = &sample_buf[SAMPLE_BUF_RIGHT_CHANNEL]; int scale = WORD_SHIFT; do { *dl++ = *sl++ << scale; *dr++ = *sr++ << scale; } while (sl < slend); src[0] = (char *)sl; src[1] = (char *)sr; } /* convert count 32-bit mono to 32-bit mono */ static void sample_input_gt_native_mono( int count, const char *src[], int32_t *dst[]) { dst[0] = dst[1] = (int32_t *)src[0]; src[0] = (char *)(dst[0] + count); } /* convert count 32-bit interleaved stereo to 32-bit noninterleaved stereo */ static void sample_input_gt_native_i_stereo( int count, const char *src[], int32_t *dst[]) { const int32_t *s = (int32_t *)src[0]; const int32_t * const send = s + 2*count; int32_t *dl = dst[0] = &sample_buf[SAMPLE_BUF_LEFT_CHANNEL]; int32_t *dr = dst[1] = &sample_buf[SAMPLE_BUF_RIGHT_CHANNEL]; do { *dl++ = *s++; *dr++ = *s++; } while (s < send); src[0] = (char *)send; } /* convert 32 bit-noninterleaved stereo to 32-bit noninterleaved stereo */ static void sample_input_gt_native_ni_stereo( int count, const char *src[], int32_t *dst[]) { dst[0] = (int32_t *)src[0]; dst[1] = (int32_t *)src[1]; src[0] = (char *)(dst[0] + count); src[1] = (char *)(dst[1] + count); } /** * sample_input_new_format() * * set the to-native sample conversion function based on dsp sample parameters * * !DSPPARAMSYNC * needs syncing with changes to the following dsp parameters: * * dsp->stereo_mode (A/V) * * dsp->sample_depth (A/V) */ static void sample_input_new_format(void) { static const sample_input_fn_type sample_input_functions[] = { [SAMPLE_INPUT_LE_NATIVE_MONO] = sample_input_lte_native_mono, [SAMPLE_INPUT_LE_NATIVE_I_STEREO] = sample_input_lte_native_i_stereo, [SAMPLE_INPUT_LE_NATIVE_NI_STEREO] = sample_input_lte_native_ni_stereo, [SAMPLE_INPUT_GT_NATIVE_MONO] = sample_input_gt_native_mono, [SAMPLE_INPUT_GT_NATIVE_I_STEREO] = sample_input_gt_native_i_stereo, [SAMPLE_INPUT_GT_NATIVE_NI_STEREO] = sample_input_gt_native_ni_stereo, }; int convert = dsp->stereo_mode; if (dsp->sample_depth > NATIVE_DEPTH) convert += SAMPLE_INPUT_GT_NATIVE_1ST_INDEX; dsp->input_samples = sample_input_functions[convert]; } #ifndef DSP_HAVE_ASM_SAMPLE_OUTPUT_MONO /* write mono internal format to output format */ static void sample_output_mono(int count, struct dsp_data *data, int32_t *src[], int16_t *dst) { const int32_t *s0 = src[0]; const int scale = data->output_scale; do { int32_t lr = clip_sample_16(*s0++ >> scale); *dst++ = lr; *dst++ = lr; } while (--count > 0); } #endif /* DSP_HAVE_ASM_SAMPLE_OUTPUT_MONO */ /* write stereo internal format to output format */ #ifndef DSP_HAVE_ASM_SAMPLE_OUTPUT_STEREO static void sample_output_stereo(int count, struct dsp_data *data, int32_t *src[], int16_t *dst) { const int32_t *s0 = src[0]; const int32_t *s1 = src[1]; const int scale = data->output_scale; do { *dst++ = clip_sample_16(*s0++ >> scale); *dst++ = clip_sample_16(*s1++ >> scale); } while (--count > 0); } #endif /* DSP_HAVE_ASM_SAMPLE_OUTPUT_STEREO */ /** * The "dither" code to convert the 24-bit samples produced by libmad was * taken from the coolplayer project - coolplayer.sourceforge.net * * This function handles mono and stereo outputs. */ static void sample_output_dithered(int count, struct dsp_data *data, int32_t *src[], int16_t *dst) { const int32_t mask = dither_mask; const int32_t bias = dither_bias; const int scale = data->output_scale; const int32_t min = data->clip_min; const int32_t max = data->clip_max; const int32_t range = max - min; int ch; int16_t *d; for (ch = 0; ch < dsp->data.num_channels; ch++) { struct dither_data * const dither = &dither_data[ch]; int32_t *s = src[ch]; int i; for (i = 0, d = &dst[ch]; i < count; i++, s++, d += 2) { int32_t output, sample; int32_t random; /* Noise shape and bias */ sample = *s; 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; output += (random & mask) - (dither->random & mask); dither->random = random; /* Clip */ int32_t c = output - min; if ((uint32_t)c > (uint32_t)range) { output -= c; if (c > 0) { output += range; if (sample > max) sample = max; } else if (sample < min) { sample = min; } } output &= ~mask; /* Error feedback */ dither->error[0] = sample - output; /* Quantize */ *d = output >> scale; } } if (dsp->data.num_channels == 2) return; /* Have to duplicate left samples into the right channel since pcm buffer and hardware is interleaved stereo */ d = &dst[0]; do { int16_t s = *d++; *d++ = s; } while (--count > 0); } /** * sample_output_new_format() * * set the from-native to ouput sample conversion routine * * !DSPPARAMSYNC * needs syncing with changes to the following dsp parameters: * * dsp->stereo_mode (A/V) * * dither_enabled (A) */ static void sample_output_new_format(void) { static const sample_output_fn_type sample_output_functions[] = { sample_output_mono, sample_output_stereo, sample_output_dithered, sample_output_dithered }; int out = dsp->data.num_channels - 1; if (dsp == audio_dsp && dither_enabled) out += 2; dsp->output_samples = sample_output_functions[out]; } /** * Linear interpolation resampling that introduces a one sample delay because * of our inability to look into the future at the end of a frame. */ #ifndef DSP_HAVE_ASM_RESAMPLING static int dsp_downsample(int count, struct dsp_data *data, int32_t *src[], int32_t *dst[]) { int ch = data->num_channels - 1; uint32_t delta = data->resample_data.delta; uint32_t phase, pos; int32_t *d; /* Rolled channel loop actually showed slightly faster. */ do { /* Just initialize things and not worry too much about the relatively * uncommon case of not being able to spit out a sample for the frame. */ int32_t *s = src[ch]; int32_t last = data->resample_data.last_sample[ch]; data->resample_data.last_sample[ch] = s[count - 1]; d = dst[ch]; phase = data->resample_data.phase; pos = phase >> 16; /* Do we need last sample of previous frame for interpolation? */ if (pos > 0) last = s[pos - 1]; while (pos < (uint32_t)count) { *d++ = last + FRACMUL((phase & 0xffff) << 15, s[pos] - last); phase += delta; pos = phase >> 16; last = s[pos - 1]; } } while (--ch >= 0); /* Wrap phase accumulator back to start of next frame. */ data->resample_data.phase = phase - (count << 16); return d - dst[0]; } static int dsp_upsample(int count, struct dsp_data *data, int32_t *src[], int32_t *dst[]) { int ch = data->num_channels - 1; uint32_t delta = data->resample_data.delta; uint32_t phase, pos; int32_t *d; /* Rolled channel loop actually showed slightly faster. */ do { /* Should always be able to output a sample for a ratio up to RESAMPLE_BUF_COUNT / SAMPLE_BUF_COUNT. */ int32_t *s = src[ch]; int32_t last = data->resample_data.last_sample[ch]; data->resample_data.last_sample[ch] = s[count - 1]; d = dst[ch]; phase = data->resample_data.phase; pos = phase >> 16; while (pos == 0) { *d++ = last + FRACMUL((phase & 0xffff) << 15, s[0] - last); phase += delta; pos = phase >> 16; } while (pos < (uint32_t)count) { last = s[pos - 1]; *d++ = last + FRACMUL((phase & 0xffff) << 15, s[pos] - last); phase += delta; pos = phase >> 16; } } while (--ch >= 0); /* Wrap phase accumulator back to start of next frame. */ data->resample_data.phase = phase & 0xffff; return d - dst[0]; } #endif /* DSP_HAVE_ASM_RESAMPLING */ static void resampler_new_delta(void) { dsp->data.resample_data.delta = (unsigned long) dsp->frequency * 65536LL / NATIVE_FREQUENCY; if (dsp->frequency == NATIVE_FREQUENCY) { /* NOTE: If fully glitch-free transistions from no resampling to resampling are desired, last_sample history should be maintained even when not resampling. */ dsp->resample = NULL; dsp->data.resample_data.phase = 0; dsp->data.resample_data.last_sample[0] = 0; dsp->data.resample_data.last_sample[1] = 0; } else if (dsp->frequency < NATIVE_FREQUENCY) dsp->resample = dsp_upsample; else dsp->resample = dsp_downsample; } /* 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(int count, int32_t *src[]) { int32_t *dst[2] = { &resample_buf[RESAMPLE_BUF_LEFT_CHANNEL], &resample_buf[RESAMPLE_BUF_RIGHT_CHANNEL], }; count = dsp->resample(count, &dsp->data, src, dst); src[0] = dst[0]; src[1] = dst[dsp->data.num_channels - 1]; return count; } static void dither_init(void) { /* Voice codec should not reset the audio codec's dither data */ if (dsp != audio_dsp) return; memset(dither_data, 0, sizeof (dither_data)); dither_bias = (1L << (dsp->frac_bits - NATIVE_DEPTH)); dither_mask = (1L << (dsp->frac_bits + 1 - NATIVE_DEPTH)) - 1; } void dsp_dither_enable(bool enable) { /* Be sure audio dsp is current to set correct function */ struct dsp_config *old_dsp = switch_dsp(audio_dsp); dither_enabled = enable; sample_output_new_format(); switch_dsp(old_dsp); } /* Applies crossfeed to the stereo signal in src. * Crossfeed is a process where listening over speakers is simulated. This * is good for old hard panned stereo records, which might be quite fatiguing * to listen to on headphones with no crossfeed. */ #ifndef DSP_HAVE_ASM_CROSSFEED static void apply_crossfeed(int count, int32_t *buf[]) { int32_t *hist_l = &crossfeed_data.history[0]; int32_t *hist_r = &crossfeed_data.history[2]; int32_t *delay = &crossfeed_data.delay[0][0]; int32_t *coefs = &crossfeed_data.coefs[0]; int32_t gain = crossfeed_data.gain; int32_t *di = crossfeed_data.index; int32_t acc; int32_t left, right; int i; for (i = 0; i < count; i++) { left = buf[0][i]; right = buf[1][i]; /* Filter delayed sample from left speaker */ ACC_INIT(acc, *di, coefs[0]); ACC(acc, hist_l[0], coefs[1]); ACC(acc, hist_l[1], coefs[2]); /* Save filter history for left speaker */ hist_l[1] = GET_ACC(acc); hist_l[0] = *di; *di++ = left; /* Filter delayed sample from right speaker */ ACC_INIT(acc, *di, coefs[0]); ACC(acc, hist_r[0], coefs[1]); ACC(acc, hist_r[1], coefs[2]); /* Save filter history for right speaker */ hist_r[1] = GET_ACC(acc); hist_r[0] = *di; *di++ = right; /* Now add the attenuated direct sound and write to outputs */ buf[0][i] = FRACMUL(left, gain) + hist_r[1]; buf[1][i] = FRACMUL(right, gain) + hist_l[1]; /* Wrap delay line index if bigger than delay line size */ if (di >= delay + 13*2) di = delay; } /* Write back local copies of data we've modified */ crossfeed_data.index = di; } #endif /* DSP_HAVE_ASM_CROSSFEED */ /** * dsp_set_crossfeed(bool enable) * * !DSPPARAMSYNC * needs syncing with changes to the following dsp parameters: * * dsp->stereo_mode (A) */ void dsp_set_crossfeed(bool enable) { crossfeed_enabled = enable; audio_dsp->apply_crossfeed = (enable && audio_dsp->data.num_channels > 1) ? apply_crossfeed : NULL; } void dsp_set_crossfeed_direct_gain(int gain) { crossfeed_data.gain = get_replaygain_int(gain * -10) << 7; /* If gain is negative, the calculation overflowed and we need to clamp */ if (crossfeed_data.gain < 0) crossfeed_data.gain = 0x7fffffff; } /* Both gains should be below 0 dB (when inverted) */ void dsp_set_crossfeed_cross_params(long lf_gain, long hf_gain, long cutoff) { int32_t *c = crossfeed_data.coefs; long scaler = get_replaygain_int(lf_gain * -10) << 7; cutoff = 0xffffffff/NATIVE_FREQUENCY*cutoff; hf_gain -= lf_gain; /* Divide cutoff by sqrt(10^(-hf_gain/20)) to place cutoff at the -3 dB * point instead of shelf midpoint. This is for compatibility with the old * crossfeed shelf filter and should be removed if crossfeed settings are * ever made incompatible for any other good reason. */ cutoff = DIV64(cutoff, get_replaygain_int(-hf_gain*5), 24); filter_shelf_coefs(cutoff, -hf_gain, false, c); /* Scale coefs by LF gain and shift them to s0.31 format. We have no gains * over 1 and can do this safely */ c[0] = FRACMUL_SHL(c[0], scaler, 4); c[1] = FRACMUL_SHL(c[1], scaler, 4); c[2] <<= 4; } /* Apply a constant gain to the samples (e.g., for ReplayGain). * Note that this must be called before the resampler. */ #ifndef DSP_HAVE_ASM_APPLY_GAIN static void dsp_apply_gain(int count, struct dsp_data *data, int32_t *buf[]) { const int32_t gain = data->gain; int ch = data->num_channels - 1; do { int32_t *s = buf[ch]; int32_t *d = buf[ch]; int32_t samp = *s++; int i = 0; do { FRACMUL_8_LOOP(samp, gain, s, d); } while (++i < count); } while (--ch >= 0); } #endif /* DSP_HAVE_ASM_APPLY_GAIN */ /* Combine all gains to a global gain. */ static void set_gain(struct dsp_config *dsp) { dsp->data.gain = DEFAULT_GAIN; /* Replay gain not relevant to voice */ if (dsp == audio_dsp && replaygain) { dsp->data.gain = replaygain; } if (eq_enabled && eq_precut) { dsp->data.gain = (long) (((int64_t) dsp->data.gain * eq_precut) >> 24); } if (dsp->data.gain == DEFAULT_GAIN) { dsp->data.gain = 0; } else { dsp->data.gain >>= 1; } dsp->apply_gain = dsp->data.gain != 0 ? dsp_apply_gain : NULL; } /** * Use to enable the equalizer. * * @param enable true to enable the equalizer */ void dsp_set_eq(bool enable) { eq_enabled = enable; } /** * Update the amount to cut the audio before applying the equalizer. * * @param precut to apply in decibels (multiplied by 10) */ void dsp_set_eq_precut(int precut) { eq_precut = get_replaygain_int(precut * -10); set_gain(audio_dsp); set_gain(voice_dsp); /* For EQ precut */ } /** * Synchronize the equalizer filter coefficients with the global settings. * * @param band the equalizer band to synchronize */ void dsp_set_eq_coefs(int band) { const int *setting; long gain; unsigned long cutoff, q; /* Adjust setting pointer to the band we actually want to change */ setting = &global_settings.eq_band0_cutoff + (band * 3); /* Convert user settings to format required by coef generator functions */ cutoff = 0xffffffff / NATIVE_FREQUENCY * (*setting++); q = *setting++; gain = *setting++; if (q == 0) q = 1; /* NOTE: The coef functions assume the EMAC unit is in fractional mode, which it should be, since we're executed from the main thread. */ /* Assume a band is disabled if the gain is zero */ if (gain == 0) { eq_data.enabled[band] = 0; } else { if (band == 0) eq_ls_coefs(cutoff, q, gain, eq_data.filters[band].coefs); else if (band == 4) eq_hs_coefs(cutoff, q, gain, eq_data.filters[band].coefs); else eq_pk_coefs(cutoff, q, gain, eq_data.filters[band].coefs); eq_data.enabled[band] = 1; } } /* Apply EQ filters to those bands that have got it switched on. */ static void eq_process(int count, int32_t *buf[]) { static const int shifts[] = { EQ_SHELF_SHIFT, /* low shelf */ EQ_PEAK_SHIFT, /* peaking */ EQ_PEAK_SHIFT, /* peaking */ EQ_PEAK_SHIFT, /* peaking */ EQ_SHELF_SHIFT, /* high shelf */ }; unsigned int channels = dsp->data.num_channels; int i; /* 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]) continue; eq_filter(buf, &eq_data.filters[i], count, channels, shifts[i]); } } void dsp_set_stereo_width(int value) { long width, straight, cross; width = value * 0x7fffff / 100; if (value <= 100) { straight = (0x7fffff + width) / 2; cross = straight - width; } else { /* straight = (1 + width) / (2 * width) */ straight = ((int64_t)(0x7fffff + width) << 22) / width; cross = straight - 0x7fffff; } dsp_sw_gain = straight << 8; dsp_sw_cross = cross << 8; } #if CONFIG_CODEC == SWCODEC #ifdef HAVE_SW_TONE_CONTROLS static void set_tone_controls(void) { filter_bishelf_coefs(0xffffffff/NATIVE_FREQUENCY*200, 0xffffffff/NATIVE_FREQUENCY*3500, bass, treble, -prescale, tone_filter.coefs); } #endif int dsp_callback(int msg, intptr_t param) { switch (msg) { #ifdef HAVE_SW_TONE_CONTROLS case DSP_CALLBACK_SET_PRESCALE: prescale = param; set_tone_controls(); break; /* prescaler is always set after calling any of these, so we wait with * calculating coefs until the above case is hit. */ case DSP_CALLBACK_SET_BASS: bass = param; break; case DSP_CALLBACK_SET_TREBLE: treble = param; #endif case DSP_CALLBACK_SET_CHANNEL_CONFIG: dsp_set_channel_config(param); break; case DSP_CALLBACK_SET_STEREO_WIDTH: dsp_set_stereo_width(param); break; default: break; } return 0; } #endif /** * Implements the different channel configurations and stereo width. */ /* SOUND_CHAN_STEREO mode is a noop so has no function - just outline one for * completeness. */ #if 0 static void channels_process_sound_chan_stereo(int count, int32_t *buf[]) { /* The channels are each just themselves */ (void)count; (void)buf; } #endif #ifndef DSP_HAVE_ASM_SOUND_CHAN_MONO static void channels_process_sound_chan_mono(int count, int32_t *buf[]) { int32_t *sl = buf[0], *sr = buf[1]; do { int32_t lr = *sl/2 + *sr/2; *sl++ = lr; *sr++ = lr; } while (--count > 0); } #endif /* DSP_HAVE_ASM_SOUND_CHAN_MONO */ #ifndef DSP_HAVE_ASM_SOUND_CHAN_CUSTOM static void channels_process_sound_chan_custom(int count, int32_t *buf[]) { const int32_t gain = dsp_sw_gain; const int32_t cross = dsp_sw_cross; int32_t *sl = buf[0], *sr = buf[1]; do { int32_t l = *sl; int32_t r = *sr; *sl++ = FRACMUL(l, gain) + FRACMUL(r, cross); *sr++ = FRACMUL(r, gain) + FRACMUL(l, cross); } while (--count > 0); } #endif /* DSP_HAVE_ASM_SOUND_CHAN_CUSTOM */ static void channels_process_sound_chan_mono_left(int count, int32_t *buf[]) { /* Just copy over the other channel */ memcpy(buf[1], buf[0], count * sizeof (*buf)); } static void channels_process_sound_chan_mono_right(int count, int32_t *buf[]) { /* Just copy over the other channel */ memcpy(buf[0], buf[1], count * sizeof (*buf)); } #ifndef DSP_HAVE_ASM_SOUND_CHAN_KARAOKE static void channels_process_sound_chan_karaoke(int count, int32_t *buf[]) { int32_t *sl = buf[0], *sr = buf[1]; do { int32_t ch = *sl/2 - *sr/2; *sl++ = ch; *sr++ = -ch; } while (--count > 0); } #endif /* DSP_HAVE_ASM_SOUND_CHAN_KARAOKE */ void dsp_set_channel_config(int value) { static const channels_process_fn_type channels_process_functions[] = { /* SOUND_CHAN_STEREO = All-purpose index for no channel processing */ [SOUND_CHAN_STEREO] = NULL, [SOUND_CHAN_MONO] = channels_process_sound_chan_mono, [SOUND_CHAN_CUSTOM] = channels_process_sound_chan_custom, [SOUND_CHAN_MONO_LEFT] = channels_process_sound_chan_mono_left, [SOUND_CHAN_MONO_RIGHT] = channels_process_sound_chan_mono_right, [SOUND_CHAN_KARAOKE] = channels_process_sound_chan_karaoke, }; if ((unsigned)value >= ARRAYLEN(channels_process_functions) || audio_dsp->stereo_mode == STEREO_MONO) value = SOUND_CHAN_STEREO; /* This doesn't apply to voice */ channels_mode = value; audio_dsp->channels_process = channels_process_functions[value]; } /* Process and convert src audio to dst based on the DSP configuration, * reading count number of audio samples. dst is assumed to be large * enough; use dsp_output_count() to get the required number. src is an * array of pointers; for mono and interleaved stereo, it contains one * pointer to the start of the audio data and the other is ignored; for * non-interleaved stereo, it contains two pointers, one for each audio * channel. Returns number of bytes written to dst. */ int dsp_process(char *dst, const char *src[], int count) { int32_t *tmp[2]; int written = 0; 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 if (new_gain) dsp_set_replaygain(); /* Gain has changed */ /* Testing function pointers for NULL is preferred since the pointer will be preloaded to be used for the call if not. */ while (count > 0) { samples = MIN(SAMPLE_BUF_COUNT/2, count); count -= samples; dsp->input_samples(samples, src, tmp); if (dsp->apply_gain) dsp->apply_gain(samples, &dsp->data, tmp); if (dsp->resample && (samples = resample(samples, tmp)) <= 0) break; /* I'm pretty sure we're downsampling here */ if (dsp->apply_crossfeed) dsp->apply_crossfeed(samples, tmp); /* TODO: EQ and tone controls need separate structs for audio and voice * DSP processing thanks to filter history. isn't really audible now, but * might be the day we start handling voice more delicately. Planned * changes may well run all relevent channels through the same EQ so * perhaps not. */ if (eq_enabled) eq_process(samples, tmp); #ifdef HAVE_SW_TONE_CONTROLS if ((bass | treble) != 0) eq_filter(tmp, &tone_filter, samples, dsp->data.num_channels, FILTER_BISHELF_SHIFT); #endif if (dsp->channels_process) dsp->channels_process(samples, tmp); dsp->output_samples(samples, &dsp->data, tmp, (int16_t *)dst); written += samples; dst += samples * sizeof (int16_t) * 2; yield(); } #if defined(CPU_COLDFIRE) && !defined(SIMULATOR) /* set old macsr again */ coldfire_set_macsr(old_macsr); #endif return written; } /* Given count number of input samples, calculate the maximum number of * samples 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 */ int dsp_output_count(int count) { if (dsp->resample) { count = (int)(((unsigned long)count * NATIVE_FREQUENCY + (dsp->frequency - 1)) / dsp->frequency); } /* Now we have the resampled sample count which must not exceed * RESAMPLE_BUF_COUNT/2 to avoid resample buffer overflow. One * must call dsp_input_count() to get the correct input sample * count. */ if (count > RESAMPLE_BUF_COUNT/2) count = RESAMPLE_BUF_COUNT/2; return count; } /* Given count output samples, calculate number of input samples * that would be consumed in order to fill the output buffer. */ int dsp_input_count(int count) { /* count is now the number of resampled input samples. Convert to original input samples. */ if (dsp->resample) { /* Use the real resampling delta = * dsp->frequency * 65536 / NATIVE_FREQUENCY, and * round towards zero to avoid buffer overflows. */ count = (int)(((unsigned long)count * dsp->data.resample_data.delta) >> 16); } return count; } int dsp_stereo_mode(void) { return dsp->stereo_mode; } bool dsp_configure(int setting, intptr_t value) { void set_gain_var(long *var, long value) { /* Voice shouldn't mess with these */ if (dsp == audio_dsp) { *var = value; new_gain = true; } } void update_functions(void) { sample_input_new_format(); sample_output_new_format(); if (dsp == audio_dsp) dsp_set_crossfeed(crossfeed_enabled); } switch (setting) { case DSP_SWITCH_CODEC: if ((uintptr_t)value <= 1) switch_dsp(&dsp_conf[value]); break; case DSP_SET_FREQUENCY: memset(&dsp->data.resample_data, 0, sizeof (dsp->data.resample_data)); /* Fall through!!! */ case DSP_SWITCH_FREQUENCY: dsp->codec_frequency = (value == 0) ? NATIVE_FREQUENCY : value; /* Account for playback speed adjustment when setting dsp->frequency if we're called from the main audio thread. Voice UI thread should not need this feature. */ if (dsp == audio_dsp) dsp->frequency = pitch_ratio * dsp->codec_frequency / 1000; else dsp->frequency = dsp->codec_frequency; resampler_new_delta(); break; case DSP_SET_SAMPLE_DEPTH: dsp->sample_depth = value; if (dsp->sample_depth <= NATIVE_DEPTH) { dsp->frac_bits = WORD_FRACBITS; dsp->sample_bytes = sizeof (int16_t); /* samples are 16 bits */ dsp->data.clip_max = ((1 << WORD_FRACBITS) - 1); dsp->data.clip_min = -((1 << WORD_FRACBITS)); } else { dsp->frac_bits = value; dsp->sample_bytes = sizeof (int32_t); /* samples are 32 bits */ dsp->data.clip_max = (1 << value) - 1; dsp->data.clip_min = -(1 << value); } dsp->data.output_scale = dsp->frac_bits + 1 - NATIVE_DEPTH; sample_input_new_format(); dither_init(); break; case DSP_SET_STEREO_MODE: dsp->stereo_mode = value; dsp->data.num_channels = value == STEREO_MONO ? 1 : 2; update_functions(); break; case DSP_RESET: dsp->stereo_mode = STEREO_NONINTERLEAVED; dsp->data.num_channels = 2; dsp->sample_depth = NATIVE_DEPTH; dsp->frac_bits = WORD_FRACBITS; dsp->sample_bytes = sizeof (int16_t); dsp->data.output_scale = dsp->frac_bits + 1 - NATIVE_DEPTH; dsp->data.clip_max = ((1 << WORD_FRACBITS) - 1); dsp->data.clip_min = -((1 << WORD_FRACBITS)); dsp->codec_frequency = dsp->frequency = NATIVE_FREQUENCY; if (dsp == audio_dsp) { track_gain = 0; album_gain = 0; track_peak = 0; album_peak = 0; new_gain = true; } update_functions(); break; case DSP_FLUSH: memset(&dsp->data.resample_data, 0, sizeof (dsp->data.resample_data)); resampler_new_delta(); dither_init(); break; case DSP_SET_TRACK_GAIN: set_gain_var(&track_gain, value); break; case DSP_SET_ALBUM_GAIN: set_gain_var(&album_gain, value); break; case DSP_SET_TRACK_PEAK: set_gain_var(&track_peak, value); break; case DSP_SET_ALBUM_PEAK: set_gain_var(&album_peak, value); break; default: return 0; } return 1; } void dsp_set_replaygain(void) { long gain = 0; new_gain = false; if (global_settings.replaygain || global_settings.replaygain_noclip) { bool track_mode = get_replaygain_mode(track_gain != 0, album_gain != 0) == REPLAYGAIN_TRACK; long peak = (track_mode || !album_peak) ? track_peak : album_peak; if (global_settings.replaygain) { gain = (track_mode || !album_gain) ? track_gain : 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_GAIN; } if (global_settings.replaygain_noclip && (peak != 0) && ((((int64_t) gain * peak) >> 24) >= DEFAULT_GAIN)) { gain = (((int64_t) DEFAULT_GAIN << 24) / peak); } if (gain == DEFAULT_GAIN) { /* Nothing to do, disable processing. */ gain = 0; } } /* Store in S8.23 format to simplify calculations. */ replaygain = gain; set_gain(audio_dsp); }