From cf19ba5599b1cba212705ddb22166acf25eca83c Mon Sep 17 00:00:00 2001 From: Jeffrey Goode Date: Fri, 25 Sep 2009 15:46:38 +0000 Subject: Replace limiter with dynamic range compressor git-svn-id: svn://svn.rockbox.org/rockbox/trunk@22832 a1c6a512-1295-4272-9138-f99709370657 --- apps/dsp.c | 638 +++++++++++++++++++++++-------------------------------------- 1 file changed, 236 insertions(+), 402 deletions(-) (limited to 'apps/dsp.c') diff --git a/apps/dsp.c b/apps/dsp.c index e7a6a9182a..a5ceecb048 100644 --- a/apps/dsp.c +++ b/apps/dsp.c @@ -137,6 +137,15 @@ struct eq_state /* 10ch */ }; +struct compressor_menu +{ + int threshold; /* dB - from menu */ + int ratio; /* from menu */ + int gain; /* dB - from menu */ + bool soft_knee; /* 0 = hard knee, 1 = soft knee */ + int release; /* samples - from menu */ +}; + /* Include header with defines which functions are implemented in assembly code for the target */ #include @@ -171,7 +180,6 @@ struct dsp_config int32_t tdspeed_percent; /* Speed% * PITCH_SPEED_PRECISION */ bool tdspeed_active; /* Timestretch is in use */ int frac_bits; - long limiter_preamp; /* limiter amp gain in S7.24 format */ #ifdef HAVE_SW_TONE_CONTROLS /* Filter struct for software bass/treble controls */ struct eqfilter tone_filter; @@ -187,7 +195,7 @@ struct dsp_config channels_process_fn_type apply_crossfeed; channels_process_fn_type eq_process; channels_process_fn_type channels_process; - return_fn_type limiter_process; + return_fn_type compressor_process; }; /* General DSP config */ @@ -253,58 +261,17 @@ static int32_t *resample_buf; #define RESAMPLE_BUF_LEFT_CHANNEL 0 #define RESAMPLE_BUF_RIGHT_CHANNEL (sample_buf_count/2 * RESAMPLE_RATIO) -/* limiter */ -/* MAX_COUNT is largest possible sample count in limiter_process. This is - needed in case time stretch makes the count in dsp_process larger than - the limiter buffer. */ -#define MAX_COUNT MAX(SMALL_SAMPLE_BUF_COUNT * RESAMPLE_RATIO / 2, LIMITER_BUFFER_SIZE) -static int count_adjust; -static bool limiter_buffer_active; -static bool limiter_buffer_full; -static bool limiter_buffer_emptying; -static int32_t limiter_buffer[2][LIMITER_BUFFER_SIZE] IBSS_ATTR; -static int32_t *start_lim_buf[2] IBSS_ATTR, - *end_lim_buf[2] IBSS_ATTR; -static uint16_t lim_buf_peak[LIMITER_BUFFER_SIZE] IBSS_ATTR; -static uint16_t *start_peak IBSS_ATTR, - *end_peak IBSS_ATTR; -static uint16_t out_buf_peak[MAX_COUNT]; -static uint16_t *out_buf_peak_index IBSS_ATTR; -static uint16_t release_peak IBSS_ATTR; -static int32_t in_samp IBSS_ATTR, - samp0 IBSS_ATTR; - -static void reset_limiter_buffer(struct dsp_config *dsp); -static int limiter_buffer_count(bool buf_count); -static int limiter_process(int count, int32_t *buf[]); -static uint16_t get_peak_value(int32_t sample); - - /* The clip_steps array essentially stores the results of fp_factor from - * 0 to 12 dB, in 48 equal steps, in S3.28 format. */ -const long clip_steps[49] ICONST_ATTR = { 0x10000000, - 0x10779AFA, 0x10F2B409, 0x1171654C, 0x11F3C9A0, 0x1279FCAD, - 0x13041AE9, 0x139241A2, 0x14248EF9, 0x14BB21F9, 0x15561A92, - 0x15F599A0, 0x1699C0F9, 0x1742B36B, 0x17F094CE, 0x18A38A01, - 0x195BB8F9, 0x1A1948C5, 0x1ADC619B, 0x1BA52CDC, 0x1C73D51D, - 0x1D488632, 0x1E236D3A, 0x1F04B8A1, 0x1FEC982C, 0x20DB3D0E, - 0x21D0D9E2, 0x22CDA2BE, 0x23D1CD41, 0x24DD9099, 0x25F12590, - 0x270CC693, 0x2830AFD3, 0x295D1F37, 0x2A925471, 0x2BD0911F, - 0x2D1818B3, 0x2E6930AD, 0x2FC42095, 0x312931EC, 0x3298B072, - 0x3412EA24, 0x35982F3A, 0x3728D22E, 0x38C52808, 0x3A6D8847, - 0x3C224CD9, 0x3DE3D264, 0x3FB2783F}; -/* The gain_steps array essentially stores the results of fp_factor from - * 0 to -12 dB, in 48 equal steps, in S3.28 format. */ -const long gain_steps[49] ICONST_ATTR = { 0x10000000, - 0xF8BC9C0, 0xF1ADF94, 0xEAD2988, 0xE429058, 0xDDAFD68, - 0xD765AC1, 0xD149309, 0xCB59186, 0xC594210, 0xBFF9112, - 0xBA86B88, 0xB53BEF5, 0xB017965, 0xAB18964, 0xA63DDFE, - 0xA1866BA, 0x9CF1397, 0x987D507, 0x9429BEE, 0x8FF599E, - 0x8BDFFD3, 0x87E80B0, 0x840CEBE, 0x804DCE8, 0x7CA9E76, - 0x792070E, 0x75B0AB0, 0x7259DB2, 0x6F1B4BF, 0x6BF44D5, - 0x68E4342, 0x65EA5A0, 0x63061D6, 0x6036E15, 0x5D7C0D3, - 0x5AD50CE, 0x5841505, 0x55C04B8, 0x535176A, 0x50F44D9, - 0x4EA84FE, 0x4C6D00E, 0x4A41E78, 0x48268DF, 0x461A81C, - 0x441D53E, 0x422E985, 0x404DE62}; +/* compressor */ +/* MAX_COUNT is largest possible sample count in compressor_process */ +#define MAX_COUNT (SMALL_SAMPLE_BUF_COUNT * RESAMPLE_RATIO / 2) +static struct compressor_menu c_menu; +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[65] IBSS_ATTR; /* S7.24 format */ +static int32_t gain_buffer[MAX_COUNT] IBSS_ATTR; +static int32_t release_gain IBSS_ATTR; + +static int compressor_process(int count, int32_t *buf[]); /* Clip sample to signed 16 bit range */ @@ -944,13 +911,6 @@ static void set_gain(struct dsp_config *dsp) dsp->data.gain = fp_mul(dsp->data.gain, eq_precut, 24); } - /* only preamp for the limiter if limiter is active and sample depth - * allows safe pre-amping (12 dB is OK with 29 or less frac bits) */ - if ((dsp->limiter_preamp) && (dsp->frac_bits <= 29)) - { - dsp->data.gain = fp_mul(dsp->data.gain, dsp->limiter_preamp, 24); - } - #ifdef HAVE_SW_VOLUME_CONTROL if (global_settings.volume < SW_VOLUME_MAX || global_settings.volume > SW_VOLUME_MIN) @@ -1308,8 +1268,8 @@ int dsp_process(struct dsp_config *dsp, char *dst, const char *src[], int count) if (dsp->channels_process) dsp->channels_process(chunk, t2); - if (dsp->limiter_process) - chunk = dsp->limiter_process(chunk, t2); + if (dsp->compressor_process) + chunk = dsp->compressor_process(chunk, t2); dsp->output_samples(chunk, &dsp->data, (const int32_t **)t2, (int16_t *)dst); @@ -1358,15 +1318,6 @@ int dsp_output_count(struct dsp_config *dsp, int count) if (count > RESAMPLE_BUF_RIGHT_CHANNEL) count = RESAMPLE_BUF_RIGHT_CHANNEL; - /* If the limiter buffer is filling, some or all samples will - * be captured by it, so expect fewer samples coming out. */ - if (limiter_buffer_active && !limiter_buffer_full) - { - int empty_space = limiter_buffer_count(false); - count_adjust = MIN(empty_space, count); - count -= count_adjust; - } - return count; } @@ -1375,13 +1326,6 @@ int dsp_output_count(struct dsp_config *dsp, int count) */ int dsp_input_count(struct dsp_config *dsp, int count) { - /* If the limiter buffer is filling, the output count was - * adjusted downward. This adjusts it back so that input - * count is not affected. - */ - if (limiter_buffer_active && !limiter_buffer_full) - count += count_adjust; - /* count is now the number of resampled input samples. Convert to original input samples. */ if (dsp->resample) @@ -1499,7 +1443,8 @@ intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value) dsp_update_functions(dsp); resampler_new_delta(dsp); tdspeed_setup(dsp); - reset_limiter_buffer(dsp); + if (dsp == &AUDIO_DSP) + release_gain = (1 << 24); break; case DSP_FLUSH: @@ -1508,7 +1453,8 @@ intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value) resampler_new_delta(dsp); dither_init(dsp); tdspeed_setup(dsp); - reset_limiter_buffer(dsp); + if (dsp == &AUDIO_DSP) + release_gain = (1 << 24); break; case DSP_SET_TRACK_GAIN: @@ -1588,369 +1534,257 @@ void dsp_set_replaygain(void) set_gain(&AUDIO_DSP); } -/** RESET THE LIMITER BUFFER - * Force the limiter buffer to its initial state and discard - * any samples held there. */ -static void reset_limiter_buffer(struct dsp_config *dsp) -{ - if (dsp == &AUDIO_DSP) +/** SET COMPRESSOR + * Called by the menu system to configure the compressor process */ +void dsp_set_compressor(int c_threshold, int c_ratio, int c_gain, + int c_knee, int c_release) +{ + bool changed = false; + bool active = (c_threshold < 0); + const int comp_ratio[] = {2, 4, 6, 10, 0}; + int new_ratio = comp_ratio[c_ratio]; + bool new_knee = (c_knee == 1); + int new_release = c_release * NATIVE_FREQUENCY / 1000; + + if (c_menu.threshold != c_threshold) { - int i; - logf(" reset_limiter_buffer"); - for (i = 0; i < 2; i++) - start_lim_buf[i] = end_lim_buf[i] = limiter_buffer[i]; - start_peak = end_peak = lim_buf_peak; - limiter_buffer_full = false; - limiter_buffer_emptying = false; - release_peak = 0; + changed = true; + c_menu.threshold = c_threshold; + logf(" Compressor Threshold: %d dB\tEnabled: %s", + c_menu.threshold, active ? "Yes" : "No"); } -} -/** OPERATE THE LIMITER BUFFER - * Handle all samples entering or exiting the limiter buffer. */ -static inline int set_limiter_buffer(int count, int32_t *buf[]) -{ - int32_t *in_buf[] = {buf[0], buf[1]}, - *out_buf[] = {buf[0], buf[1]}; - int empty_space, i, out_count; - const long clip_max = AUDIO_DSP.data.clip_max; - const int ch = AUDIO_DSP.data.num_channels - 1; - out_buf_peak_index = out_buf_peak; - - if (limiter_buffer_emptying) - /** EMPTY THE BUFFER - * since the empty flag has been set, assume no inbound samples and - return all samples in the limiter buffer to the outbound buffer */ + if (c_menu.ratio != new_ratio) { - count = limiter_buffer_count(true); - out_count = count; - logf(" Emptying limiter buffer: %d", count); - while (count-- > 0) - { - for (i = 0; i <= ch; i++) - { - /* move samples in limiter buffer to output buffer */ - *out_buf[i]++ = *start_lim_buf[i]++; - if (start_lim_buf[i] == &limiter_buffer[i][LIMITER_BUFFER_SIZE]) - start_lim_buf[i] = limiter_buffer[i]; - /* move limiter buffer peak values to output peak values */ - if (i == 0) - { - *out_buf_peak_index++ = *start_peak++; - if (start_peak == &lim_buf_peak[LIMITER_BUFFER_SIZE]) - start_peak = lim_buf_peak; - } - } - } - limiter_buffer_full = false; - limiter_buffer_emptying = false; + changed = true; + c_menu.ratio = new_ratio; + if (c_menu.ratio) + logf(" Compressor Ratio: %d:1", c_menu.ratio); + else + logf(" Compressor Ratio: Limit"); } - else /* limiter buffer NOT emptying */ + + if (c_menu.gain != c_gain) + { + changed = true; + c_menu.gain = c_gain; + if (c_menu.gain >= 0) + logf(" Compressor Makeup Gain: %d dB", c_menu.gain); + else + logf(" Compressor Makeup Gain: Auto"); + } + + if (c_menu.soft_knee != new_knee) { - if (count <= 0) return 0; + changed = true; + c_menu.soft_knee = new_knee; + logf(" Compressor Knee: %s", c_menu.soft_knee==1?"Soft":"Hard"); + } + + if (c_menu.release != new_release) + { + changed = true; + c_menu.release = new_release; + logf(" Compressor Release: %d", c_menu.release); + } + + if (changed && active) + { + /* configure variables for compressor operation */ + int i; + 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}; - empty_space = limiter_buffer_count(false); + struct curve_point + { + int32_t db; /* S15.16 format */ + int32_t offset; /* S15.16 format */ + } db_curve[4]; - if (empty_space > 0) - /** FILL BUFFER - * use as many inbound samples as necessary to fill the buffer */ + /** 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 */ + db_curve[1].db = c_menu.threshold << 16; + db_curve[1].offset = 0; + if (c_menu.soft_knee) { - /* don't try to fill with more samples than available */ - if (empty_space > count) - empty_space = count; - logf(" Filling limiter buffer: %d", empty_space); - while (empty_space-- > 0) - { - for (i = 0; i <= ch; i++) - { - /* put inbound samples in the limiter buffer */ - in_samp = *in_buf[i]++; - *end_lim_buf[i]++ = in_samp; - if (end_lim_buf[i] == &limiter_buffer[i][LIMITER_BUFFER_SIZE]) - end_lim_buf[i] = limiter_buffer[i]; - if (in_samp < 0) /* make positive for comparison */ - in_samp = -in_samp - 1; - if (in_samp <= clip_max) - in_samp = 0; /* disregard if not clipped */ - if (i == 0) - samp0 = in_samp; - if (i == ch) - { - /* assign peak value for each inbound sample pair */ - *end_peak++ = ((samp0 > 0) || (in_samp > 0)) ? - get_peak_value(MAX(samp0, in_samp)) : 0; - if (end_peak == &lim_buf_peak[LIMITER_BUFFER_SIZE]) - end_peak = lim_buf_peak; - } - } - count--; - } - /* after buffer fills, the remaining inbound samples are cycled */ + /* bottom of knee is 3dB below the threshold for soft knee*/ + db_curve[0].db = db_curve[1].db - (3 << 16); + db_curve[0].offset = 0; + /* top of knee is 3dB above the threshold for soft knee */ + db_curve[2].db = db_curve[1].db + (3 << 16); + if (c_menu.ratio) + /* offset = -3db * (ratio - 1) / ratio */ + db_curve[2].offset = (int32_t)((long long)(-3 << 16) + * (c_menu.ratio - 1) / c_menu.ratio); + else + /* offset = -3db for hard limit */ + db_curve[2].offset = (-3 << 16); } - - limiter_buffer_full = (end_lim_buf[0] == start_lim_buf[0]); - out_count = count; - - /** CYCLE BUFFER - * return buffered samples and backfill limiter buffer with new ones. - * The buffer is always full when cycling. */ - while (count-- > 0) + else { - for (i = 0; i <= ch; i++) - { - /* copy incoming sample */ - in_samp = *in_buf[i]++; - /* put limiter buffer sample into outbound buffer */ - *out_buf[i]++ = *start_lim_buf[i]++; - /* put incoming sample on the end of the limiter buffer */ - *end_lim_buf[i]++ = in_samp; - /* ring buffer pointer wrap */ - if (start_lim_buf[i] == &limiter_buffer[i][LIMITER_BUFFER_SIZE]) - start_lim_buf[i] = limiter_buffer[i]; - if (end_lim_buf[i] == &limiter_buffer[i][LIMITER_BUFFER_SIZE]) - end_lim_buf[i] = limiter_buffer[i]; - if (in_samp < 0) /* make positive for comparison */ - in_samp = -in_samp - 1; - if (in_samp <= clip_max) - in_samp = 0; /* disregard if not clipped */ - if (i == 0) - { - samp0 = in_samp; - /* assign outgoing sample its associated peak value */ - *out_buf_peak_index++ = *start_peak++; - if (start_peak == &lim_buf_peak[LIMITER_BUFFER_SIZE]) - start_peak = lim_buf_peak; - } - if (i == ch) - { - /* assign peak value for each inbound sample pair */ - *end_peak++ = ((samp0 > 0) || (in_samp > 0)) ? - get_peak_value(MAX(samp0, in_samp)) : 0; - if (end_peak == &lim_buf_peak[LIMITER_BUFFER_SIZE]) - end_peak = lim_buf_peak; - } - } + /* bottom of knee is at the threshold for hard knee */ + db_curve[0].db = c_menu.threshold << 16; + db_curve[0].offset = 0; + /* top of knee is at the threshold for hard knee */ + db_curve[2].db = c_menu.threshold << 16; + db_curve[2].offset = 0; } - } - - return out_count; -} - -/** RETURN LIMITER BUFFER COUNT - * If argument is true, returns number of samples in the buffer, - * otherwise, returns empty space remaining */ -static int limiter_buffer_count(bool buf_count) -{ - int count; - if (limiter_buffer_full) - count = LIMITER_BUFFER_SIZE; - else if (end_lim_buf[0] >= start_lim_buf[0]) - count = (end_lim_buf[0] - start_lim_buf[0]); - else - count = (end_lim_buf[0] - start_lim_buf[0]) + LIMITER_BUFFER_SIZE; - return buf_count ? count : (LIMITER_BUFFER_SIZE - count); -} - -/** FLUSH THE LIMITER BUFFER - * Empties the limiter buffer into the buffer pointed to by the argument - * and returns the number of samples in that buffer */ -int dsp_flush_limiter_buffer(char *dest) -{ - if ((!limiter_buffer_active) || (limiter_buffer_count(true) <= 0)) - return 0; - - logf(" dsp_flush_limiter_buffer"); - int32_t flush_buf[2][LIMITER_BUFFER_SIZE]; - int32_t *src[2] = {flush_buf[0], flush_buf[1]}; - - limiter_buffer_emptying = true; - int count = limiter_process(0, src); - AUDIO_DSP.output_samples(count, &AUDIO_DSP.data, - (const int32_t **)src, (int16_t *)dest); - return count; -} - -/** GET PEAK VALUE - * Return a small value representing how much the sample is clipped. This - * should only be called if a sample is actually clipped. Sample is a - * positive value. - */ -static uint16_t get_peak_value(int32_t sample) -{ - const int frac_bits = AUDIO_DSP.frac_bits; - int mid, - hi = 48, - lo = 0; - - /* shift sample into 28 frac bit range for comparison */ - if (frac_bits > 28) - sample >>= (frac_bits - 28); - if (frac_bits < 28) - sample <<= (28 - frac_bits); - - /* if clipped out of range, return maximum value */ - if (sample >= clip_steps[48]) - return 48 * 90; - - /* find amount of sample clipping on the table */ - do - { - mid = (hi + lo) / 2; - if (sample < clip_steps[mid]) - hi = mid; - else if (sample > clip_steps[mid]) - lo = mid; + /* 0db input is also max offset point (most compression) */ + db_curve[3].db = 0; + if (c_menu.ratio) + /* offset = threshold * (ratio - 1) / ratio */ + db_curve[3].offset = (int32_t)((long long)(c_menu.threshold << 16) + * (c_menu.ratio - 1) / c_menu.ratio); else - return mid * 90; - } - while (hi > (lo + 1)); - - /* interpolate linearly between steps (less accurate but faster) */ - return ((hi-1) * 90) + (((sample - clip_steps[hi-1]) * 90) / - (clip_steps[hi] - clip_steps[hi-1])); -} - -/** SET LIMITER - * Called by the menu system to configure the limiter process */ -void dsp_set_limiter(int limiter_level) -{ - if (limiter_level > 0) - { - if (!limiter_buffer_active) + /* offset = threshold for hard limit */ + db_curve[3].offset = (c_menu.threshold << 16); + + /* Now set up the comp_curve table with compression offsets in the form + of gain factors in S7.24 format */ + comp_curve[0] = (1 << 24); + for (i = 1; i < 64; i++) { - /* enable limiter process */ - AUDIO_DSP.limiter_process = limiter_process; - limiter_buffer_active = true; + int32_t this_db = -db[i]; + /* no compression below the knee */ + if (this_db <= db_curve[0].db) + comp_curve[i] = (1 << 24); + + /* if soft knee and below top of knee, interpolate along soft knee slope */ + else if (c_menu.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((this_db - db_curve[1].db), + -db_curve[1].db, 16), db_curve[3].offset, 16), 16) << 8; } - /* limiter preamp is a gain factor in S7.24 format */ - long old_preamp = AUDIO_DSP.limiter_preamp; - long new_preamp = fp_factor((((long)limiter_level << 24) / 10), 24); - if (old_preamp != new_preamp) + comp_curve[64] = fp_factor(db_curve[3].offset, 16) << 8; + + logf("\n *** Compression Offsets ***"); + for (i = 0; i <= 3; i++) { - AUDIO_DSP.limiter_preamp = new_preamp; - set_gain(&AUDIO_DSP); - logf(" Limiter enable: Yes\tLimiter amp: %.8f", - (float)AUDIO_DSP.limiter_preamp / (1 << 24)); + logf("Curve[%d]: db: % .1f\toffset: % .4f", i, (float)db_curve[i].db / (1 << 16), + (float)db_curve[i].offset / (1 << 16)); } - } - else - { - /* disable limiter process*/ - if (limiter_buffer_active) + + logf("\nGain factors:"); + for (i = 1; i <= 64; i++) { - AUDIO_DSP.limiter_preamp = (1 << 24); - set_gain(&AUDIO_DSP); - /* pcmbuf_flush_limiter_buffer(); */ - limiter_buffer_active = false; - AUDIO_DSP.limiter_process = NULL; - reset_limiter_buffer(&AUDIO_DSP); - logf(" Limiter enable: No\tLimiter amp: %.8f", - (float)AUDIO_DSP.limiter_preamp / (1 << 24)); + debugf("%02d: %.6f ", i, (float)comp_curve[i] / (1 << 24)); + if (i % 4 == 0) debugf("\n"); } + + /* if using auto peak, then makeup gain is max offset - .1dB headroom */ + int32_t db_makeup = (c_menu.gain == -1) ? + -(db_curve[3].offset) - 0x199A : c_menu.gain << 16; + comp_makeup_gain = fp_factor(db_makeup, 16) << 8; + logf("Makeup gain:\t%.6f", (float)comp_makeup_gain / (1 << 24)); + + /* calculate per-sample gain change a rate of 10db over release time */ + comp_rel_slope = 0xAF0BB2 / c_menu.release; + logf("Release slope:\t%.6f", (float)comp_rel_slope / (1 << 24)); + + release_gain = (1 << 24); } + + /* enable/disable the compressor */ + AUDIO_DSP.compressor_process = active ? compressor_process : NULL; } -/** LIMITER PROCESS - * Checks pre-amplified signal for clipped samples and smoothly reduces gain - * around the clipped samples using a preset attack/release schedule. +/** 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 int limiter_process(int count, int32_t *buf[]) +static inline int32_t get_compression_gain(int32_t sample) { - /* Limiter process passes through if limiter buffer isn't active, or the - * sample depth is too large for safe pre-amping */ - if ((!limiter_buffer_active) || (AUDIO_DSP.frac_bits > 29)) - return count; + const int frac_bits = AUDIO_DSP.frac_bits; - count = set_limiter_buffer(count, buf); + /* sample must be positive */ + if (sample < 0) + sample = -sample - 1; + + /* shift sample into 22 frac bit range */ + if (frac_bits > 22) + sample >>= (frac_bits - 22); + if (frac_bits < 22) + sample <<= (22 - frac_bits); - if (count <= 0) - return 0; + /* index is 6 MSB, rem is 16 LSB */ + int index = sample >> 16; + int rem = (sample & 0xFFFF) << 8; - const int attack_slope = 15; /* 15:1 ratio between attack and release */ - const int buffer_count = limiter_buffer_count(true); + /* interpolate from the compression curve */ + return comp_curve[index] + (int32_t)FRACMUL_SHL((comp_curve[index + 1] + - comp_curve[index]), rem, 7); +} + +/** COMPRESSOR PROCESS + * Changes the gain of the samples according to the compressor curve + */ +static int compressor_process(int count, int32_t *buf[]) +{ + const int num_chan = AUDIO_DSP.data.num_channels; + const int32_t fp_one = (1 << 24); + int32_t sample_gain, /* S7.24 format */ + this_gain; /* S7.24 format */ int i, ch; - uint16_t max_peak = 0, - gain_peak, - gain_rem; - long gain; - /* step through limiter buffer in reverse order, in order to find the - * appropriate max_peak for modifying the output buffer */ - for (i = buffer_count - 1; i >= 0; i--) - { - const uint16_t peak_i = lim_buf_peak[(start_peak - lim_buf_peak + i) % - LIMITER_BUFFER_SIZE]; - /* if no attack slope, nothing to do */ - if ((peak_i == 0) && (max_peak == 0)) continue; - /* if new peak, start attack slope */ - if (peak_i >= max_peak) - { - max_peak = peak_i; - } - /* keep sloping */ - else - { - if (max_peak > attack_slope) - max_peak -= attack_slope; - else - max_peak = 0; - } - } - /* step through output buffer the same way, but this time modifying peak - * values to create a smooth attack slope. */ - for (i = count - 1; i >= 0; i--) - { - /* if no attack slope, nothing to do */ - if ((out_buf_peak[i] == 0) && (max_peak == 0)) continue; - /* if new peak, start attack slope */ - if (out_buf_peak[i] >= max_peak) - { - max_peak = out_buf_peak[i]; - } - /* keep sloping */ - else - { - if (max_peak > attack_slope) - max_peak -= attack_slope; - else - max_peak = 0; - out_buf_peak[i] = max_peak; - } - } - /* Now step forward through the output buffer, and modify the peak values + /* Step forward through the output buffer, and modify the offset values * to establish a smooth, slow release slope.*/ for (i = 0; i < count; i++) { - /* if no release slope, nothing to do */ - if ((out_buf_peak[i] == 0) && (release_peak == 0)) continue; - /* if new peak, start release slope */ - if (out_buf_peak[i] >= release_peak) + sample_gain = fp_one; + for (ch = 0; ch < num_chan; ch++) { - release_peak = out_buf_peak[i]; + this_gain = get_compression_gain(buf[ch][i]); + if (this_gain < sample_gain) + sample_gain = this_gain; } - /* keep sloping */ + /* if no release slope, only apply makeup gain */ + if ((sample_gain == fp_one) && (release_gain == fp_one)) + gain_buffer[i] = comp_makeup_gain; else { - release_peak--; - out_buf_peak[i] = release_peak; + /* if larger offset, start release slope */ + if (sample_gain <= release_gain) + release_gain = sample_gain; + else /* keep sloping */ + { + if (release_gain < (fp_one - comp_rel_slope)) + release_gain += comp_rel_slope; + else + release_gain = fp_one; + } + /* store offset with release and also apply makeup gain */ + if ((release_gain == fp_one) && (comp_makeup_gain == fp_one)) + gain_buffer[i] = fp_one; + else + gain_buffer[i] = FRACMUL_SHL(release_gain, comp_makeup_gain, 7); } } - /* Implement the limiter: adjust gain of the outbound samples by the gain - * amounts in the gain steps array corresponding to the peak values. */ + + /* Implement the compressor: apply those gain factors to the output + * buffer samples */ + for (i = 0; i < count; i++) { - if (out_buf_peak[i] > 0) + if (gain_buffer[i] != fp_one) { - gain_peak = (out_buf_peak[i] + 1) / 90; - gain_rem = (out_buf_peak[i] + 1) % 90; - gain = gain_steps[gain_peak]; - if ((gain_peak < 48) && (gain_rem > 0)) - gain -= gain_rem * ((gain_steps[gain_peak] - - gain_steps[gain_peak + 1]) / 90); - for (ch = 0; ch < AUDIO_DSP.data.num_channels; ch++) - buf[ch][i] = FRACMUL_SHL(buf[ch][i], gain, 3); + for (ch = 0; ch < num_chan; ch++) + buf[ch][i] = FRACMUL_SHL(buf[ch][i], gain_buffer[i], 7); } } return count; -} +} -- cgit v1.2.3