From 1ab9d14c77adc241ff1b126f216dbac8dd34e3fc Mon Sep 17 00:00:00 2001 From: Michael Sevakis Date: Wed, 8 Feb 2012 14:55:37 -0500 Subject: Move to compressor out of dsp.c and into its own source to reduce DSP clutter. A bit of a rough job for the moment but all works. Change-Id: Id40852e0dec99caee02f943d0da8a1cdc16f022a --- apps/SOURCES | 1 + apps/compressor.c | 363 ++++++++++++++++++++++++++++++++++++++++++++++++++++ apps/compressor.h | 29 +++++ apps/dsp.c | 371 +++--------------------------------------------------- apps/dsp.h | 38 ++++++ 5 files changed, 446 insertions(+), 356 deletions(-) create mode 100644 apps/compressor.c create mode 100644 apps/compressor.h diff --git a/apps/SOURCES b/apps/SOURCES index 53a67fd307..e1990217ca 100644 --- a/apps/SOURCES +++ b/apps/SOURCES @@ -169,6 +169,7 @@ codec_thread.c playback.c codecs.c dsp.c +compressor.c #ifndef HAVE_HARDWARE_BEEP beep.c #endif diff --git a/apps/compressor.c b/apps/compressor.c new file mode 100644 index 0000000000..3a8d52e4da --- /dev/null +++ b/apps/compressor.c @@ -0,0 +1,363 @@ +/*************************************************************************** + * __________ __ ___. + * Open \______ \ ____ ____ | | _\_ |__ _______ ___ + * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / + * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < + * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ + * \/ \/ \/ \/ \/ + * $Id$ + * + * Copyright (C) 2009 Jeffrey Goode + * + * 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 software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY + * KIND, either express or implied. + * + ****************************************************************************/ +#include "config.h" +#include "fixedpoint.h" +#include "fracmul.h" +#include "settings.h" +#include "dsp.h" +#include "compressor.h" + +/* Define LOGF_ENABLE to enable logf output in this file */ +/*#define LOGF_ENABLE*/ +#include "logf.h" + +static int32_t comp_rel_slope IBSS_ATTR; /* S7.24 format */ +static int32_t comp_makeup_gain IBSS_ATTR; /* S7.24 format */ +static int32_t comp_curve[66] IBSS_ATTR; /* S7.24 format */ +static int32_t release_gain IBSS_ATTR; /* S7.24 format */ + +#define UNITY (1L << 24) /* unity gain in S7.24 format */ + +/** COMPRESSOR UPDATE + * Called via the menu system to configure the compressor process */ +bool compressor_update(void) +{ + static int curr_set[5]; + int new_set[5] = { + global_settings.compressor_threshold, + global_settings.compressor_makeup_gain, + global_settings.compressor_ratio, + global_settings.compressor_knee, + global_settings.compressor_release_time}; + + /* make menu values useful */ + int threshold = new_set[0]; + bool auto_gain = (new_set[1] == 1); + const int comp_ratios[] = {2, 4, 6, 10, 0}; + int ratio = comp_ratios[new_set[2]]; + bool soft_knee = (new_set[3] == 1); + int release = new_set[4] * NATIVE_FREQUENCY / 1000; + + bool changed = false; + bool active = (threshold < 0); + + for (int i = 0; i < 5; i++) + { + if (curr_set[i] != new_set[i]) + { + changed = true; + curr_set[i] = new_set[i]; + +#if defined(ROCKBOX_HAS_LOGF) && defined(LOGF_ENABLE) + switch (i) + { + case 0: + logf(" Compressor Threshold: %d dB\tEnabled: %s", + threshold, active ? "Yes" : "No"); + break; + case 1: + logf(" Compressor Makeup Gain: %s", + auto_gain ? "Auto" : "Off"); + break; + case 2: + if (ratio) + { logf(" Compressor Ratio: %d:1", ratio); } + else + { logf(" Compressor Ratio: Limit"); } + break; + case 3: + logf(" Compressor Knee: %s", soft_knee?"Soft":"Hard"); + break; + case 4: + logf(" Compressor Release: %d", release); + break; + } +#endif + } + } + + if (changed && active) + { + /* configure variables for compressor operation */ + static const int32_t db[] = { + /* positive db equivalents in S15.16 format */ + 0x000000, 0x241FA4, 0x1E1A5E, 0x1A94C8, + 0x181518, 0x1624EA, 0x148F82, 0x1338BD, + 0x120FD2, 0x1109EB, 0x101FA4, 0x0F4BB6, + 0x0E8A3C, 0x0DD840, 0x0D3377, 0x0C9A0E, + 0x0C0A8C, 0x0B83BE, 0x0B04A5, 0x0A8C6C, + 0x0A1A5E, 0x09ADE1, 0x094670, 0x08E398, + 0x0884F6, 0x082A30, 0x07D2FA, 0x077F0F, + 0x072E31, 0x06E02A, 0x0694C8, 0x064BDF, + 0x060546, 0x05C0DA, 0x057E78, 0x053E03, + 0x04FF5F, 0x04C273, 0x048726, 0x044D64, + 0x041518, 0x03DE30, 0x03A89B, 0x037448, + 0x03412A, 0x030F32, 0x02DE52, 0x02AE80, + 0x027FB0, 0x0251D6, 0x0224EA, 0x01F8E2, + 0x01CDB4, 0x01A359, 0x0179C9, 0x0150FC, + 0x0128EB, 0x010190, 0x00DAE4, 0x00B4E1, + 0x008F82, 0x006AC1, 0x004699, 0x002305}; + + struct curve_point + { + int32_t db; /* S15.16 format */ + int32_t offset; /* S15.16 format */ + } db_curve[5]; + + /** Set up the shape of the compression curve first as decibel + values */ + /* db_curve[0] = bottom of knee + [1] = threshold + [2] = top of knee + [3] = 0 db input + [4] = ~+12db input (2 bits clipping overhead) */ + + db_curve[1].db = threshold << 16; + if (soft_knee) + { + /* bottom of knee is 3dB below the threshold for soft knee*/ + db_curve[0].db = db_curve[1].db - (3 << 16); + /* top of knee is 3dB above the threshold for soft knee */ + db_curve[2].db = db_curve[1].db + (3 << 16); + if (ratio) + /* offset = -3db * (ratio - 1) / ratio */ + db_curve[2].offset = (int32_t)((long long)(-3 << 16) + * (ratio - 1) / ratio); + else + /* offset = -3db for hard limit */ + db_curve[2].offset = (-3 << 16); + } + else + { + /* bottom of knee is at the threshold for hard knee */ + db_curve[0].db = threshold << 16; + /* top of knee is at the threshold for hard knee */ + db_curve[2].db = threshold << 16; + db_curve[2].offset = 0; + } + + /* Calculate 0db and ~+12db offsets */ + db_curve[4].db = 0xC0A8C; /* db of 2 bits clipping */ + if (ratio) + { + /* offset = threshold * (ratio - 1) / ratio */ + db_curve[3].offset = (int32_t)((long long)(threshold << 16) + * (ratio - 1) / ratio); + db_curve[4].offset = (int32_t)((long long)-db_curve[4].db + * (ratio - 1) / ratio) + db_curve[3].offset; + } + else + { + /* offset = threshold for hard limit */ + db_curve[3].offset = (threshold << 16); + db_curve[4].offset = -db_curve[4].db + db_curve[3].offset; + } + + /** Now set up the comp_curve table with compression offsets in the + form of gain factors in S7.24 format */ + /* comp_curve[0] is 0 (-infinity db) input */ + comp_curve[0] = UNITY; + /* comp_curve[1 to 63] are intermediate compression values + corresponding to the 6 MSB of the input values of a non-clipped + signal */ + for (int i = 1; i < 64; i++) + { + /* db constants are stored as positive numbers; + make them negative here */ + int32_t this_db = -db[i]; + + /* no compression below the knee */ + if (this_db <= db_curve[0].db) + comp_curve[i] = UNITY; + + /* if soft knee and below top of knee, + interpolate along soft knee slope */ + else if (soft_knee && (this_db <= db_curve[2].db)) + comp_curve[i] = fp_factor(fp_mul( + ((this_db - db_curve[0].db) / 6), + db_curve[2].offset, 16), 16) << 8; + + /* interpolate along ratio slope above the knee */ + else + comp_curve[i] = fp_factor(fp_mul( + fp_div((db_curve[1].db - this_db), db_curve[1].db, 16), + db_curve[3].offset, 16), 16) << 8; + } + /* comp_curve[64] is the compression level of a maximum level, + non-clipped signal */ + comp_curve[64] = fp_factor(db_curve[3].offset, 16) << 8; + + /* comp_curve[65] is the compression level of a maximum level, + clipped signal */ + comp_curve[65] = fp_factor(db_curve[4].offset, 16) << 8; + +#if defined(ROCKBOX_HAS_LOGF) && defined(LOGF_ENABLE) + logf("\n *** Compression Offsets ***"); + /* some settings for display only, not used in calculations */ + db_curve[0].offset = 0; + db_curve[1].offset = 0; + db_curve[3].db = 0; + + for (int i = 0; i <= 4; i++) + { + logf("Curve[%d]: db: % 6.2f\toffset: % 6.2f", i, + (float)db_curve[i].db / (1 << 16), + (float)db_curve[i].offset / (1 << 16)); + } + + logf("\nGain factors:"); + for (int i = 1; i <= 65; i++) + { + debugf("%02d: %.6f ", i, (float)comp_curve[i] / UNITY); + if (i % 4 == 0) debugf("\n"); + } + debugf("\n"); +#endif + + /* if using auto peak, then makeup gain is max offset - + .1dB headroom */ + comp_makeup_gain = auto_gain ? + fp_factor(-(db_curve[3].offset) - 0x199A, 16) << 8 : UNITY; + logf("Makeup gain:\t%.6f", (float)comp_makeup_gain / UNITY); + + /* calculate per-sample gain change a rate of 10db over release time + */ + comp_rel_slope = 0xAF0BB2 / release; + logf("Release slope:\t%.6f", (float)comp_rel_slope / UNITY); + + release_gain = UNITY; + } + + return active; +} + +/** GET COMPRESSION GAIN + * Returns the required gain factor in S7.24 format in order to compress the + * sample in accordance with the compression curve. Always 1 or less. + */ +static inline int32_t get_compression_gain(struct dsp_data *data, + int32_t sample) +{ + const int frac_bits_offset = data->frac_bits - 15; + + /* sample must be positive */ + if (sample < 0) + sample = -(sample + 1); + + /* shift sample into 15 frac bit range */ + if (frac_bits_offset > 0) + sample >>= frac_bits_offset; + if (frac_bits_offset < 0) + sample <<= -frac_bits_offset; + + /* normal case: sample isn't clipped */ + if (sample < (1 << 15)) + { + /* index is 6 MSB, rem is 9 LSB */ + int index = sample >> 9; + int32_t rem = (sample & 0x1FF) << 22; + + /* interpolate from the compression curve: + higher gain - ((rem / (1 << 31)) * (higher gain - lower gain)) */ + return comp_curve[index] - (FRACMUL(rem, + (comp_curve[index] - comp_curve[index + 1]))); + } + /* sample is somewhat clipped, up to 2 bits of overhead */ + if (sample < (1 << 17)) + { + /* straight interpolation: + higher gain - ((clipped portion of sample * 4/3 + / (1 << 31)) * (higher gain - lower gain)) */ + return comp_curve[64] - (FRACMUL(((sample - (1 << 15)) / 3) << 16, + (comp_curve[64] - comp_curve[65]))); + } + + /* sample is too clipped, return invalid value */ + return -1; +} + +/** COMPRESSOR PROCESS + * Changes the gain of the samples according to the compressor curve + */ +void compressor_process(int count, struct dsp_data *data, int32_t *buf[]) +{ + const int num_chan = data->num_channels; + int32_t *in_buf[2] = {buf[0], buf[1]}; + + while (count-- > 0) + { + int ch; + /* use lowest (most compressed) gain factor of the output buffer + sample pair for both samples (mono is also handled correctly here) + */ + int32_t sample_gain = UNITY; + for (ch = 0; ch < num_chan; ch++) + { + int32_t this_gain = get_compression_gain(data, *in_buf[ch]); + if (this_gain < sample_gain) + sample_gain = this_gain; + } + + /* perform release slope; skip if no compression and no release slope + */ + if ((sample_gain != UNITY) || (release_gain != UNITY)) + { + /* if larger offset than previous slope, start new release slope + */ + if ((sample_gain <= release_gain) && (sample_gain > 0)) + { + release_gain = sample_gain; + } + else + /* keep sloping towards unity gain (and ignore invalid value) */ + { + release_gain += comp_rel_slope; + if (release_gain > UNITY) + { + release_gain = UNITY; + } + } + } + + /* total gain factor is the product of release gain and makeup gain, + but avoid computation if possible */ + int32_t total_gain = ((release_gain == UNITY) ? comp_makeup_gain : + (comp_makeup_gain == UNITY) ? release_gain : + FRACMUL_SHL(release_gain, comp_makeup_gain, 7)); + + /* Implement the compressor: apply total gain factor (if any) to the + output buffer sample pair/mono sample */ + if (total_gain != UNITY) + { + for (ch = 0; ch < num_chan; ch++) + { + *in_buf[ch] = FRACMUL_SHL(total_gain, *in_buf[ch], 7); + } + } + in_buf[0]++; + in_buf[1]++; + } +} + +void compressor_reset(void) +{ + release_gain = UNITY; +} diff --git a/apps/compressor.h b/apps/compressor.h new file mode 100644 index 0000000000..6154372e05 --- /dev/null +++ b/apps/compressor.h @@ -0,0 +1,29 @@ +/*************************************************************************** + * __________ __ ___. + * Open \______ \ ____ ____ | | _\_ |__ _______ ___ + * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / + * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < + * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ + * \/ \/ \/ \/ \/ + * $Id$ + * + * Copyright (C) 2009 Jeffrey Goode + * + * 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 software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY + * KIND, either express or implied. + * + ****************************************************************************/ + +#ifndef COMPRESSOR_H +#define COMPRESSOR_H + +void compressor_process(int count, struct dsp_data *data, int32_t *buf[]); +bool compressor_update(void); +void compressor_reset(void); + +#endif /* COMPRESSOR_H */ diff --git a/apps/dsp.c b/apps/dsp.c index 00de511dd0..4017f6afc0 100644 --- a/apps/dsp.c +++ b/apps/dsp.c @@ -24,6 +24,7 @@ #include "dsp.h" #include "dsp-util.h" #include "eq.h" +#include "compressor.h" #include "kernel.h" #include "settings.h" #include "replaygain.h" @@ -66,42 +67,6 @@ enum 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 { @@ -154,7 +119,7 @@ typedef void (*channels_process_dsp_fn_type)(int count, struct dsp_data *data, struct dsp_config { - struct dsp_data data; /* Config members for use in asm routines */ + struct dsp_data data; /* Config members for use in external 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; @@ -164,7 +129,6 @@ struct dsp_config #ifdef HAVE_PITCHSCREEN bool tdspeed_active; /* Timestretch is in use */ #endif - int frac_bits; #ifdef HAVE_SW_TONE_CONTROLS /* Filter struct for software bass/treble controls */ struct eqfilter tone_filter; @@ -180,7 +144,7 @@ struct dsp_config channels_process_fn_type apply_crossfeed; channels_process_fn_type eq_process; channels_process_fn_type channels_process; - channels_process_fn_type compressor_process; + channels_process_dsp_fn_type compressor_process; }; /* General DSP config */ @@ -249,15 +213,6 @@ static int32_t *sample_buf[2] = { small_sample_buf[0], small_sample_buf[1] }; static int resample_buf_count = SMALL_RESAMPLE_BUF_COUNT; static int32_t *resample_buf[2] = { small_resample_buf[0], small_resample_buf[1] }; -/* compressor */ -static int32_t comp_rel_slope IBSS_ATTR; /* S7.24 format */ -static int32_t comp_makeup_gain IBSS_ATTR; /* S7.24 format */ -static int32_t comp_curve[66] IBSS_ATTR; /* S7.24 format */ -static int32_t release_gain IBSS_ATTR; /* S7.24 format */ -#define UNITY (1L << 24) /* unity gain in S7.24 format */ -static void compressor_process(int count, int32_t *buf[]); - - #ifdef HAVE_PITCHSCREEN int32_t sound_get_pitch(void) { @@ -813,8 +768,8 @@ static inline int resample(struct dsp_config *dsp, int count, int32_t *src[]) static void dither_init(struct dsp_config *dsp) { 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; + dither_bias = (1L << (dsp->data.frac_bits - NATIVE_DEPTH)); + dither_mask = (1L << (dsp->data.frac_bits + 1 - NATIVE_DEPTH)) - 1; } void dsp_dither_enable(bool enable) @@ -1319,7 +1274,7 @@ int dsp_process(struct dsp_config *dsp, char *dst, const char *src[], int count) dsp->channels_process(chunk, t2); if (dsp->compressor_process) - dsp->compressor_process(chunk, t2); + dsp->compressor_process(chunk, &dsp->data, t2); dsp->output_samples(chunk, &dsp->data, (const int32_t **)t2, (int16_t *)dst); @@ -1453,20 +1408,20 @@ intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value) if (dsp->sample_depth <= NATIVE_DEPTH) { - dsp->frac_bits = WORD_FRACBITS; + dsp->data.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->data.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; + dsp->data.output_scale = dsp->data.frac_bits + 1 - NATIVE_DEPTH; sample_input_new_format(dsp); dither_init(dsp); break; @@ -1484,9 +1439,9 @@ intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value) dsp->stereo_mode = STEREO_NONINTERLEAVED; dsp->data.num_channels = 2; dsp->sample_depth = NATIVE_DEPTH; - dsp->frac_bits = WORD_FRACBITS; + dsp->data.frac_bits = WORD_FRACBITS; dsp->sample_bytes = sizeof (int16_t); - dsp->data.output_scale = dsp->frac_bits + 1 - NATIVE_DEPTH; + dsp->data.output_scale = dsp->data.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; @@ -1506,7 +1461,7 @@ intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value) tdspeed_setup(dsp); #endif if (dsp == &AUDIO_DSP) - release_gain = UNITY; + compressor_reset(); break; case DSP_FLUSH: @@ -1518,7 +1473,7 @@ intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value) tdspeed_setup(dsp); #endif if (dsp == &AUDIO_DSP) - release_gain = UNITY; + compressor_reset(); break; case DSP_SET_TRACK_GAIN: @@ -1616,303 +1571,7 @@ void dsp_set_replaygain(void) * Called by the menu system to configure the compressor process */ void dsp_set_compressor(void) { - static int curr_set[5]; - int new_set[5] = { - global_settings.compressor_threshold, - global_settings.compressor_makeup_gain, - global_settings.compressor_ratio, - global_settings.compressor_knee, - global_settings.compressor_release_time}; - - /* make menu values useful */ - int threshold = new_set[0]; - bool auto_gain = (new_set[1] == 1); - const int comp_ratios[] = {2, 4, 6, 10, 0}; - int ratio = comp_ratios[new_set[2]]; - bool soft_knee = (new_set[3] == 1); - int release = new_set[4] * NATIVE_FREQUENCY / 1000; - - bool changed = false; - bool active = (threshold < 0); - - for (int i = 0; i < 5; i++) - { - if (curr_set[i] != new_set[i]) - { - changed = true; - curr_set[i] = new_set[i]; - -#if defined(ROCKBOX_HAS_LOGF) && defined(LOGF_ENABLE) - switch (i) - { - case 0: - logf(" Compressor Threshold: %d dB\tEnabled: %s", - threshold, active ? "Yes" : "No"); - break; - case 1: - logf(" Compressor Makeup Gain: %s", - auto_gain ? "Auto" : "Off"); - break; - case 2: - if (ratio) - { logf(" Compressor Ratio: %d:1", ratio); } - else - { logf(" Compressor Ratio: Limit"); } - break; - case 3: - logf(" Compressor Knee: %s", soft_knee?"Soft":"Hard"); - break; - case 4: - logf(" Compressor Release: %d", release); - break; - } -#endif - } - } - - if (changed && active) - { - /* configure variables for compressor operation */ - const int32_t db[] ={0x000000, /* positive db equivalents in S15.16 format */ - 0x241FA4, 0x1E1A5E, 0x1A94C8, 0x181518, 0x1624EA, 0x148F82, 0x1338BD, 0x120FD2, - 0x1109EB, 0x101FA4, 0x0F4BB6, 0x0E8A3C, 0x0DD840, 0x0D3377, 0x0C9A0E, 0x0C0A8C, - 0x0B83BE, 0x0B04A5, 0x0A8C6C, 0x0A1A5E, 0x09ADE1, 0x094670, 0x08E398, 0x0884F6, - 0x082A30, 0x07D2FA, 0x077F0F, 0x072E31, 0x06E02A, 0x0694C8, 0x064BDF, 0x060546, - 0x05C0DA, 0x057E78, 0x053E03, 0x04FF5F, 0x04C273, 0x048726, 0x044D64, 0x041518, - 0x03DE30, 0x03A89B, 0x037448, 0x03412A, 0x030F32, 0x02DE52, 0x02AE80, 0x027FB0, - 0x0251D6, 0x0224EA, 0x01F8E2, 0x01CDB4, 0x01A359, 0x0179C9, 0x0150FC, 0x0128EB, - 0x010190, 0x00DAE4, 0x00B4E1, 0x008F82, 0x006AC1, 0x004699, 0x002305}; - - struct curve_point - { - int32_t db; /* S15.16 format */ - int32_t offset; /* S15.16 format */ - } db_curve[5]; - - /** Set up the shape of the compression curve first as decibel values*/ - /* db_curve[0] = bottom of knee - [1] = threshold - [2] = top of knee - [3] = 0 db input - [4] = ~+12db input (2 bits clipping overhead) */ - - db_curve[1].db = threshold << 16; - if (soft_knee) - { - /* bottom of knee is 3dB below the threshold for soft knee*/ - db_curve[0].db = db_curve[1].db - (3 << 16); - /* top of knee is 3dB above the threshold for soft knee */ - db_curve[2].db = db_curve[1].db + (3 << 16); - if (ratio) - /* offset = -3db * (ratio - 1) / ratio */ - db_curve[2].offset = (int32_t)((long long)(-3 << 16) - * (ratio - 1) / ratio); - else - /* offset = -3db for hard limit */ - db_curve[2].offset = (-3 << 16); - } - else - { - /* bottom of knee is at the threshold for hard knee */ - db_curve[0].db = threshold << 16; - /* top of knee is at the threshold for hard knee */ - db_curve[2].db = threshold << 16; - db_curve[2].offset = 0; - } - - /* Calculate 0db and ~+12db offsets */ - db_curve[4].db = 0xC0A8C; /* db of 2 bits clipping */ - if (ratio) - { - /* offset = threshold * (ratio - 1) / ratio */ - db_curve[3].offset = (int32_t)((long long)(threshold << 16) - * (ratio - 1) / ratio); - db_curve[4].offset = (int32_t)((long long)-db_curve[4].db - * (ratio - 1) / ratio) + db_curve[3].offset; - } - else - { - /* offset = threshold for hard limit */ - db_curve[3].offset = (threshold << 16); - db_curve[4].offset = -db_curve[4].db + db_curve[3].offset; - } - - /** Now set up the comp_curve table with compression offsets in the form - of gain factors in S7.24 format */ - /* comp_curve[0] is 0 (-infinity db) input */ - comp_curve[0] = UNITY; - /* comp_curve[1 to 63] are intermediate compression values corresponding - to the 6 MSB of the input values of a non-clipped signal */ - for (int i = 1; i < 64; i++) - { - /* db constants are stored as positive numbers; - make them negative here */ - int32_t this_db = -db[i]; - - /* no compression below the knee */ - if (this_db <= db_curve[0].db) - comp_curve[i] = UNITY; - - /* if soft knee and below top of knee, - interpolate along soft knee slope */ - else if (soft_knee && (this_db <= db_curve[2].db)) - comp_curve[i] = fp_factor(fp_mul( - ((this_db - db_curve[0].db) / 6), - db_curve[2].offset, 16), 16) << 8; - - /* interpolate along ratio slope above the knee */ - else - comp_curve[i] = fp_factor(fp_mul( - fp_div((db_curve[1].db - this_db), db_curve[1].db, 16), - db_curve[3].offset, 16), 16) << 8; - } - /* comp_curve[64] is the compression level of a maximum level, - non-clipped signal */ - comp_curve[64] = fp_factor(db_curve[3].offset, 16) << 8; - - /* comp_curve[65] is the compression level of a maximum level, - clipped signal */ - comp_curve[65] = fp_factor(db_curve[4].offset, 16) << 8; - -#if defined(ROCKBOX_HAS_LOGF) && defined(LOGF_ENABLE) - logf("\n *** Compression Offsets ***"); - /* some settings for display only, not used in calculations */ - db_curve[0].offset = 0; - db_curve[1].offset = 0; - db_curve[3].db = 0; - - for (int i = 0; i <= 4; i++) - { - logf("Curve[%d]: db: % 6.2f\toffset: % 6.2f", i, - (float)db_curve[i].db / (1 << 16), - (float)db_curve[i].offset / (1 << 16)); - } - - logf("\nGain factors:"); - for (int i = 1; i <= 65; i++) - { - debugf("%02d: %.6f ", i, (float)comp_curve[i] / UNITY); - if (i % 4 == 0) debugf("\n"); - } - debugf("\n"); -#endif - - /* if using auto peak, then makeup gain is max offset - .1dB headroom */ - comp_makeup_gain = auto_gain ? - fp_factor(-(db_curve[3].offset) - 0x199A, 16) << 8 : UNITY; - logf("Makeup gain:\t%.6f", (float)comp_makeup_gain / UNITY); - - /* calculate per-sample gain change a rate of 10db over release time */ - comp_rel_slope = 0xAF0BB2 / release; - logf("Release slope:\t%.6f", (float)comp_rel_slope / UNITY); - - release_gain = UNITY; - } - /* enable/disable the compressor */ - AUDIO_DSP.compressor_process = active ? compressor_process : NULL; -} - -/** GET COMPRESSION GAIN - * Returns the required gain factor in S7.24 format in order to compress the - * sample in accordance with the compression curve. Always 1 or less. - */ -static inline int32_t get_compression_gain(int32_t sample) -{ - const int frac_bits_offset = AUDIO_DSP.frac_bits - 15; - - /* sample must be positive */ - if (sample < 0) - sample = -(sample + 1); - - /* shift sample into 15 frac bit range */ - if (frac_bits_offset > 0) - sample >>= frac_bits_offset; - if (frac_bits_offset < 0) - sample <<= -frac_bits_offset; - - /* normal case: sample isn't clipped */ - if (sample < (1 << 15)) - { - /* index is 6 MSB, rem is 9 LSB */ - int index = sample >> 9; - int32_t rem = (sample & 0x1FF) << 22; - - /* interpolate from the compression curve: - higher gain - ((rem / (1 << 31)) * (higher gain - lower gain)) */ - return comp_curve[index] - (FRACMUL(rem, - (comp_curve[index] - comp_curve[index + 1]))); - } - /* sample is somewhat clipped, up to 2 bits of overhead */ - if (sample < (1 << 17)) - { - /* straight interpolation: - higher gain - ((clipped portion of sample * 4/3 - / (1 << 31)) * (higher gain - lower gain)) */ - return comp_curve[64] - (FRACMUL(((sample - (1 << 15)) / 3) << 16, - (comp_curve[64] - comp_curve[65]))); - } - - /* sample is too clipped, return invalid value */ - return -1; -} - -/** COMPRESSOR PROCESS - * Changes the gain of the samples according to the compressor curve - */ -static void compressor_process(int count, int32_t *buf[]) -{ - const int num_chan = AUDIO_DSP.data.num_channels; - int32_t *in_buf[2] = {buf[0], buf[1]}; - - while (count-- > 0) - { - int ch; - /* use lowest (most compressed) gain factor of the output buffer - sample pair for both samples (mono is also handled correctly here) */ - int32_t sample_gain = UNITY; - for (ch = 0; ch < num_chan; ch++) - { - int32_t this_gain = get_compression_gain(*in_buf[ch]); - if (this_gain < sample_gain) - sample_gain = this_gain; - } - - /* perform release slope; skip if no compression and no release slope */ - if ((sample_gain != UNITY) || (release_gain != UNITY)) - { - /* if larger offset than previous slope, start new release slope */ - if ((sample_gain <= release_gain) && (sample_gain > 0)) - { - release_gain = sample_gain; - } - else - /* keep sloping towards unity gain (and ignore invalid value) */ - { - release_gain += comp_rel_slope; - if (release_gain > UNITY) - { - release_gain = UNITY; - } - } - } - - /* total gain factor is the product of release gain and makeup gain, - but avoid computation if possible */ - int32_t total_gain = ((release_gain == UNITY) ? comp_makeup_gain : - (comp_makeup_gain == UNITY) ? release_gain : - FRACMUL_SHL(release_gain, comp_makeup_gain, 7)); - - /* Implement the compressor: apply total gain factor (if any) to the - output buffer sample pair/mono sample */ - if (total_gain != UNITY) - { - for (ch = 0; ch < num_chan; ch++) - { - *in_buf[ch] = FRACMUL_SHL(total_gain, *in_buf[ch], 7); - } - } - in_buf[0]++; - in_buf[1]++; - } + AUDIO_DSP.compressor_process = compressor_update() ? + compressor_process : NULL; } diff --git a/apps/dsp.h b/apps/dsp.h index c42e712a5a..2a00f649f8 100644 --- a/apps/dsp.h +++ b/apps/dsp.h @@ -57,6 +57,44 @@ enum DSP_CROSSFEED }; + +/**************************************************************************** + * 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 external 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/external 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. */ + int frac_bits; /* 24h */ + /* 28h */ +}; + struct dsp_config; int dsp_process(struct dsp_config *dsp, char *dest, -- cgit v1.2.3