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Replace limiter with dynamic range compressor

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@22832 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Jeffrey Goode 2009-09-25 15:46:38 +00:00
parent b9a17dd0ce
commit cf19ba5599
21 changed files with 577 additions and 507 deletions
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644
apps/dsp.c
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@ -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 <dsp_asm.h>
@ -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;
/* 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 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};
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:
@ -1588,368 +1534,256 @@ 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)
/** 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)
{
if (dsp == &AUDIO_DSP)
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[])
if (c_menu.ratio != new_ratio)
{
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 */
{
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;
}
else /* limiter buffer NOT emptying */
{
if (count <= 0) return 0;
empty_space = limiter_buffer_count(false);
if (empty_space > 0)
/** FILL BUFFER
* use as many inbound samples as necessary to fill the buffer */
{
/* 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 */
}
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)
{
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;
}
}
}
}
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]);
changed = true;
c_menu.ratio = new_ratio;
if (c_menu.ratio)
logf(" Compressor Ratio: %d:1", c_menu.ratio);
else
count = (end_lim_buf[0] - start_lim_buf[0]) + LIMITER_BUFFER_SIZE;
return buf_count ? count : (LIMITER_BUFFER_SIZE - count);
logf(" Compressor Ratio: Limit");
}
/** 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 (c_menu.gain != c_gain)
{
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;
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");
}
/** 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.
if (c_menu.soft_knee != new_knee)
{
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};
struct curve_point
{
int32_t db; /* S15.16 format */
int32_t offset; /* S15.16 format */
} db_curve[4];
/** 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)
{
/* 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);
}
else
{
/* 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;
}
/* 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
/* 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++)
{
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;
}
comp_curve[64] = fp_factor(db_curve[3].offset, 16) << 8;
logf("\n *** Compression Offsets ***");
for (i = 0; i <= 3; i++)
{
logf("Curve[%d]: db: % .1f\toffset: % .4f", i, (float)db_curve[i].db / (1 << 16),
(float)db_curve[i].offset / (1 << 16));
}
logf("\nGain factors:");
for (i = 1; i <= 64; i++)
{
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;
}
/** 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 uint16_t get_peak_value(int32_t sample)
static inline int32_t get_compression_gain(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);
/* sample must be positive */
if (sample < 0)
sample = -sample - 1;
/* if clipped out of range, return maximum value */
if (sample >= clip_steps[48])
return 48 * 90;
/* shift sample into 22 frac bit range */
if (frac_bits > 22)
sample >>= (frac_bits - 22);
if (frac_bits < 22)
sample <<= (22 - frac_bits);
/* 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;
else
return mid * 90;
}
while (hi > (lo + 1));
/* index is 6 MSB, rem is 16 LSB */
int index = sample >> 16;
int rem = (sample & 0xFFFF) << 8;
/* 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]));
/* interpolate from the compression curve */
return comp_curve[index] + (int32_t)FRACMUL_SHL((comp_curve[index + 1]
- comp_curve[index]), rem, 7);
}
/** 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)
{
/* enable limiter process */
AUDIO_DSP.limiter_process = limiter_process;
limiter_buffer_active = true;
}
/* 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)
{
AUDIO_DSP.limiter_preamp = new_preamp;
set_gain(&AUDIO_DSP);
logf(" Limiter enable: Yes\tLimiter amp: %.8f",
(float)AUDIO_DSP.limiter_preamp / (1 << 24));
}
}
else
{
/* disable limiter process*/
if (limiter_buffer_active)
{
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));
}
}
}
/** LIMITER PROCESS
* Checks pre-amplified signal for clipped samples and smoothly reduces gain
* around the clipped samples using a preset attack/release schedule.
/** COMPRESSOR PROCESS
* Changes the gain of the samples according to the compressor curve
*/
static int limiter_process(int count, int32_t *buf[])
static int compressor_process(int count, int32_t *buf[])
{
/* 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;
count = set_limiter_buffer(count, buf);
if (count <= 0)
return 0;
const int attack_slope = 15; /* 15:1 ratio between attack and release */
const int buffer_count = limiter_buffer_count(true);
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;

7
apps/dsp.h
View file

@ -26,8 +26,7 @@
#include <stdbool.h>
#define NATIVE_FREQUENCY 44100
#define LIMITER_BUFFER_SIZE 288 /* ~6.5 ms */
#define MAX_LIMITER_GAIN 80 /* 8 dB */
enum
{
STEREO_INTERLEAVED = 0,
@ -82,7 +81,7 @@ int32_t sound_get_pitch(void);
void dsp_set_timestretch(int32_t percent);
int32_t dsp_get_timestretch(void);
int dsp_callback(int msg, intptr_t param);
int dsp_flush_limiter_buffer(char *dest);
void dsp_set_limiter(int limiter_level);
void dsp_set_compressor(int c_threshold, int c_ratio, int c_gain,
int c_knee, int c_release);
#endif

2
apps/fixedpoint.h
View file

@ -61,7 +61,7 @@ long fp_sincos(unsigned long phase, long *cos);
#define FP_NEGINF -(0x7fffffff)
/* fracbits in range 12 - 22 work well. Higher is better for
* calculating dB, lower is better for calculating ratio.
* calculating dB, lower is better for calculating factor.
*/
/* long fp_decibels(unsigned long factor, unsigned int fracbits); */
long fp_factor(long decibels, unsigned int fracbits);

6
apps/lang/dansk.lang
View file

@ -12646,15 +12646,15 @@
user: core
<source>
*: none
swcodec: "Limiter Preamp"
swcodec: "Compressor"
</source>
<dest>
*: none
swcodec: "Begrænser forforstærkning"
swcodec: "Compressor"
</dest>
<voice>
*: none
swcodec: "Begrænser for-forstærkning"
swcodec: "Compressor"
</voice>
</phrase>
<phrase>

6
apps/lang/deutsch.lang
View file

@ -12641,15 +12641,15 @@
user: core
<source>
*: none
swcodec: "Limiter Preamp"
swcodec: "Compressor"
</source>
<dest>
*: none
swcodec: "Begrenzer-Vorverstärkung"
swcodec: "Compressor"
</dest>
<voice>
*: none
swcodec: "Begrenzer-Vorverstärkung"
swcodec: "Compressor"
</voice>
</phrase>
<phrase>

227
apps/lang/english.lang
View file

@ -12707,15 +12707,15 @@
user: core
<source>
*: none
swcodec: "Limiter Preamp"
swcodec: "Compressor"
</source>
<dest>
*: none
swcodec: "Limiter Preamp"
swcodec: "Compressor"
</dest>
<voice>
*: none
swcodec: "Limiter Preamp"
swcodec: "Compressor"
</voice>
</phrase>
<phrase>
@ -12769,3 +12769,224 @@
radio: "Other"
</voice>
</phrase>
<phrase>
id: LANG_COMPRESSOR_THRESHOLD
desc: in sound settings
user: core
<source>
*: none
swcodec: "Threshold"
</source>
<dest>
*: none
swcodec: "Threshold"
</dest>
<voice>
*: none
swcodec: "Threshold"
</voice>
</phrase>
<phrase>
id: LANG_COMPRESSOR_RATIO
desc: in sound settings
user: core
<source>
*: none
swcodec: "Ratio"
</source>
<dest>
*: none
swcodec: "Ratio"
</dest>
<voice>
*: none
swcodec: "Ratio"
</voice>
</phrase>
<phrase>
id: LANG_COMPRESSOR_RATIO_2
desc: in sound settings
user: core
<source>
*: none
swcodec: "2:1"
</source>
<dest>
*: none
swcodec: "2:1"
</dest>
<voice>
*: none
swcodec: "2 to 1"
</voice>
</phrase>
<phrase>
id: LANG_COMPRESSOR_RATIO_4
desc: in sound settings
user: core
<source>
*: none
swcodec: "4:1"
</source>
<dest>
*: none
swcodec: "4:1"
</dest>
<voice>
*: none
swcodec: "4 to 1"
</voice>
</phrase>
<phrase>
id: LANG_COMPRESSOR_RATIO_6
desc: in sound settings
user: core
<source>
*: none
swcodec: "6:1"
</source>
<dest>
*: none
swcodec: "6:1"
</dest>
<voice>
*: none
swcodec: "6 to 1"
</voice>
</phrase>
<phrase>
id: LANG_COMPRESSOR_RATIO_10
desc: in sound settings
user: core
<source>
*: none
swcodec: "10:1"
</source>
<dest>
*: none
swcodec: "10:1"
</dest>
<voice>
*: none
swcodec: "10 to 1"
</voice>
</phrase>
<phrase>
id: LANG_COMPRESSOR_RATIO_LIMIT
desc: in sound settings
user: core
<source>
*: none
swcodec: "Limit"
</source>
<dest>
*: none
swcodec: "Limit"
</dest>
<voice>
*: none
swcodec: "Limit"
</voice>
</phrase>
<phrase>
id: LANG_COMPRESSOR_GAIN
desc: in sound settings
user: core
<source>
*: none
swcodec: "Makeup Gain"
</source>
<dest>
*: none
swcodec: "Makeup Gain"
</dest>
<voice>
*: none
swcodec: "Makeup Gain"
</voice>
</phrase>
<phrase>
id: LANG_AUTO
desc: in sound settings
user: core
<source>
*: none
swcodec: "Auto"
</source>
<dest>
*: none
swcodec: "Auto"
</dest>
<voice>
*: none
swcodec: "Auto"
</voice>
</phrase>
<phrase>
id: LANG_COMPRESSOR_KNEE
desc: in sound settings
user: core
<source>
*: none
swcodec: "Knee"
</source>
<dest>
*: none
swcodec: "Knee"
</dest>
<voice>
*: none
swcodec: "Knee"
</voice>
</phrase>
<phrase>
id: LANG_COMPRESSOR_HARD_KNEE
desc: in sound settings
user: core
<source>
*: none
swcodec: "Hard Knee"
</source>
<dest>
*: none
swcodec: "Hard Knee"
</dest>
<voice>
*: none
swcodec: "Hard Knee"
</voice>
</phrase>
<phrase>
id: LANG_COMPRESSOR_SOFT_KNEE
desc: in sound settings
user: core
<source>
*: none
swcodec: "Soft Knee"
</source>
<dest>
*: none
swcodec: "Soft Knee"
</dest>
<voice>
*: none
swcodec: "Soft Knee"
</voice>
</phrase>
<phrase>
id: LANG_COMPRESSOR_RELEASE
desc: in sound settings
user: core
<source>
*: none
swcodec: "Release Time"
</source>
<dest>
*: none
swcodec: "Release Time"
</dest>
<voice>
*: none
swcodec: "Release Time"
</voice>
</phrase>

6
apps/lang/francais.lang
View file

@ -12663,15 +12663,15 @@
user: core
<source>
*: none
swcodec: "Limiter Preamp"
swcodec: "Compressor"
</source>
<dest>
*: none
swcodec: "Limiteur préampli."
swcodec: "Compressor"
</dest>
<voice>
*: none
swcodec: "Limiteur préampli"
swcodec: "Compressor"
</voice>
</phrase>
<phrase>

6
apps/lang/italiano.lang
View file

@ -12636,15 +12636,15 @@
user: core
<source>
*: none
swcodec: "Limiter Preamp"
swcodec: "Compressor"
</source>
<dest>
*: none
swcodec: "Limitatore Preamp"
swcodec: "Compressor"
</dest>
<voice>
*: none
swcodec: "Limitatore Preamp"
swcodec: "Compressor"
</voice>
</phrase>
<phrase>

6
apps/lang/polski.lang
View file

@ -12644,15 +12644,15 @@
user: core
<source>
*: none
swcodec: "Limiter Preamp"
swcodec: "Compressor"
</source>
<dest>
*: none
swcodec: "Wzmacnianie cichych fragmentów"
swcodec: "Compressor"
</dest>
<voice>
*: none
swcodec: "Wzmacnianie cichych fragmentów"
swcodec: "Compressor"
</voice>
</phrase>
<phrase>

6
apps/lang/svenska.lang
View file

@ -12637,15 +12637,15 @@
user: core
<source>
*: none
swcodec: "Limiter Preamp"
swcodec: "Compressor"
</source>
<dest>
*: none
swcodec: "Limiter-förförstärkning"
swcodec: "Compressor"
</dest>
<voice>
*: none
swcodec: "Limiter-förförstärkning"
swcodec: "Compressor"
</voice>
</phrase>
<phrase>

6
apps/lang/tagalog.lang
View file

@ -12445,15 +12445,15 @@
user: core
<source>
*: none
swcodec: "Limiter Preamp"
swcodec: "Compressor"
</source>
<dest>
*: none
swcodec: "Limiter Preamp"
swcodec: "Compressor"
</dest>
<voice>
*: none
swcodec: "Limiter Preamp"
swcodec: "Compressor"
</voice>
</phrase>
<phrase>

8
apps/lang/walon.lang
View file

@ -12632,19 +12632,19 @@
</voice>
</phrase>
<phrase>
id: LANG_LIMITER
id: LANG_COMPRESSOR
desc: in sound settings
user: core
<source>
*: none
swcodec: "Limiter Preamp"
swcodec: "Compressor"
</source>
<dest>
*: none
swcodec: "Aschatrece Preamp"
swcodec: "Compressor"
</dest>
<voice>
*: none
swcodec: "Aschatrece Preamp"
swcodec: "Compressor"
</voice>
</phrase>

19
apps/menus/sound_menu.c
View file

@ -105,8 +105,21 @@ static int timestretch_callback(int action,const struct menu_item_ex *this_item)
&global_settings.timestretch_enabled, timestretch_callback);
MENUITEM_SETTING(dithering_enabled,
&global_settings.dithering_enabled, lowlatency_callback);
MENUITEM_SETTING(limiter_level,
&global_settings.limiter_level, lowlatency_callback);
/* compressor submenu */
MENUITEM_SETTING(compressor_threshold,
&global_settings.compressor_threshold, lowlatency_callback);
MENUITEM_SETTING(compressor_ratio,
&global_settings.compressor_ratio, lowlatency_callback);
MENUITEM_SETTING(compressor_gain,
&global_settings.compressor_makeup_gain, lowlatency_callback);
MENUITEM_SETTING(compressor_knee,
&global_settings.compressor_knee, lowlatency_callback);
MENUITEM_SETTING(compressor_release,
&global_settings.compressor_release_time, lowlatency_callback);
MAKE_MENU(compressor_menu,ID2P(LANG_COMPRESSOR), NULL, Icon_NOICON,
&compressor_threshold, &compressor_ratio, &compressor_gain,
&compressor_knee, &compressor_release);
#endif
#if (CONFIG_CODEC == MAS3587F) || (CONFIG_CODEC == MAS3539F)
@ -140,7 +153,7 @@ MAKE_MENU(sound_settings, ID2P(LANG_SOUND_SETTINGS), NULL, Icon_Audio,
#if CONFIG_CODEC == SWCODEC
,&crossfeed_menu, &equalizer_menu, &dithering_enabled
,&timestretch_enabled
,&limiter_level
,&compressor_menu
#endif
#if (CONFIG_CODEC == MAS3587F) || (CONFIG_CODEC == MAS3539F)
,&loudness,&avc,&superbass,&mdb_enable,&mdb_strength

24
apps/pcmbuf.c
View file

@ -1170,30 +1170,6 @@ bool pcmbuf_is_crossfade_enabled(void)
* Commit any remaining samples in the PCM buffer for playback. */
void pcmbuf_play_remainder(void)
{
pcmbuf_flush_limiter_buffer();
if (audiobuffer_fillpos)
pcmbuf_flush_fillpos();
}
/** FLUSH LIMITER BUFFER
* Empty the limiter buffer and commit its contents
* to the PCM buffer for playback. */
void pcmbuf_flush_limiter_buffer(void)
{
char *dest;
int out_count = LIMITER_BUFFER_SIZE;
/* create room at the end of the PCM buffer for any
samples that may be held back in the limiter buffer */
while ((dest = pcmbuf_request_buffer(&out_count)) == NULL)
{
cancel_cpu_boost();
sleep(1);
}
/* flush the limiter buffer into the PCM buffer */
out_count = dsp_flush_limiter_buffer(dest);
if (out_count > 0)
pcmbuf_write_complete(out_count);
}

1
apps/pcmbuf.h
View file

@ -76,6 +76,5 @@ void pcmbuf_beep(unsigned int frequency, size_t duration, int amplitude);
int pcmbuf_used_descs(void);
int pcmbuf_descs(void);
void pcmbuf_play_remainder(void);
void pcmbuf_flush_limiter_buffer(void);
#endif

1
apps/plugin.c
View file

@ -467,7 +467,6 @@ static const struct plugin_api rockbox_api = {
dsp_process,
dsp_input_count,
dsp_output_count,
dsp_flush_limiter_buffer,
#endif /* CONFIG_CODEC == SWCODEC */
/* playback control */

5
apps/plugin.h
View file

@ -133,12 +133,12 @@ void* plugin_get_buffer(size_t *buffer_size);
#define PLUGIN_MAGIC 0x526F634B /* RocK */
/* increase this every time the api struct changes */
#define PLUGIN_API_VERSION 171
#define PLUGIN_API_VERSION 172
/* update this to latest version if a change to the api struct breaks
backwards compatibility (and please take the opportunity to sort in any
new function which are "waiting" at the end of the function table) */
#define PLUGIN_MIN_API_VERSION 171
#define PLUGIN_MIN_API_VERSION 172
/* plugin return codes */
enum plugin_status {
@ -596,7 +596,6 @@ struct plugin_api {
const char *src[], int count);
int (*dsp_input_count)(struct dsp_config *dsp, int count);
int (*dsp_output_count)(struct dsp_config *dsp, int count);
int (*dsp_flush_limiter_buffer)(char *dest);
#endif /* CONFIG_CODEC == SWCODC */
/* playback control */

24
apps/plugins/test_codec.c
View file

@ -810,34 +810,10 @@ show_menu:
rb->closedir(dir);
}
/* process last samples */
if (use_dsp)
rb->dsp_flush_limiter_buffer(dspbuffer);
} else {
/* Just test the file */
res = test_track(parameter);
/* process last samples */
if (use_dsp)
{
int channels = (wavinfo.stereomode == STEREO_MONO) ? 1 : 2;
int count = rb->dsp_flush_limiter_buffer(dspbuffer);
if (channels == 1)
{
unsigned char *s = dspbuffer, *d = dspbuffer;
int c = count;
while (c-- > 0)
{
*d++ = *s++;
*d++ = *s++;
s++;
s++;
}
}
if (wavinfo.fd >= 0)
rb->write(wavinfo.fd, dspbuffer, count * 2 * channels);
}
/* Close WAV file (if there was one) */
if (wavinfo.fd >= 0) {
close_wav();

6
apps/settings.c
View file

@ -956,7 +956,11 @@ void settings_apply(bool read_disk)
dsp_dither_enable(global_settings.dithering_enabled);
dsp_timestretch_enable(global_settings.timestretch_enabled);
dsp_set_limiter(global_settings.limiter_level);
dsp_set_compressor(global_settings.compressor_threshold,
global_settings.compressor_ratio,
global_settings.compressor_makeup_gain,
global_settings.compressor_knee,
global_settings.compressor_release_time);
#endif
#ifdef HAVE_SPDIF_POWER

6
apps/settings.h
View file

@ -777,7 +777,11 @@ struct user_settings
#endif
#if CONFIG_CODEC == SWCODEC
int limiter_level;
int compressor_threshold;
int compressor_ratio;
int compressor_makeup_gain;
int compressor_knee;
int compressor_release_time;
#endif
};

56
apps/settings_list.c
View file

@ -356,6 +356,34 @@ static void crossfeed_cross_set(int val)
global_settings.crossfeed_hf_cutoff);
}
static void compressor_set(int val)
{
(void)val;
dsp_set_compressor(global_settings.compressor_threshold,
global_settings.compressor_ratio,
global_settings.compressor_makeup_gain,
global_settings.compressor_knee,
global_settings.compressor_release_time);
}
static const char* auto_formatter(char *buffer, size_t buffer_size,
int val, const char *unit)
{
if (val == -1)
return str(LANG_AUTO);
else
snprintf(buffer, buffer_size, "%d %s", val, unit);
return buffer;
}
static int32_t auto_getlang(int value, int unit)
{
if (value == -1)
return LANG_AUTO;
else
return TALK_ID(value, unit);
}
static const char* db_format(char* buffer, size_t buffer_size, int value,
const char* unit)
{
@ -1256,11 +1284,29 @@ const struct settings_list settings[] = {
OFFON_SETTING(F_SOUNDSETTING, timestretch_enabled, LANG_TIMESTRETCH, false,
"timestretch enabled", dsp_timestretch_enable),
/* limiter */
INT_SETTING_NOWRAP(F_SOUNDSETTING, limiter_level,
LANG_COMPRESSOR, 0,
"limiter level", UNIT_DB, 0, MAX_LIMITER_GAIN,
5, db_format, get_dec_talkid, dsp_set_limiter),
/* compressor */
INT_SETTING_NOWRAP(F_SOUNDSETTING, compressor_threshold,
LANG_COMPRESSOR_THRESHOLD, 0,
"compressor threshold", UNIT_DB, 0, -24,
-3, formatter_unit_0_is_off, getlang_unit_0_is_off, compressor_set),
CHOICE_SETTING(F_SOUNDSETTING|F_NO_WRAP, compressor_ratio,
LANG_COMPRESSOR_RATIO, 1, "compressor ratio",
"2:1,4:1,6:1,10:1,limit", compressor_set, 5,
ID2P(LANG_COMPRESSOR_RATIO_2), ID2P(LANG_COMPRESSOR_RATIO_4),
ID2P(LANG_COMPRESSOR_RATIO_6), ID2P(LANG_COMPRESSOR_RATIO_10),
ID2P(LANG_COMPRESSOR_RATIO_LIMIT)),
INT_SETTING_NOWRAP(F_SOUNDSETTING, compressor_makeup_gain,
LANG_COMPRESSOR_GAIN, -1,
"compressor makeup gain", UNIT_DB, -1, 20,
1, auto_formatter, auto_getlang, compressor_set),
CHOICE_SETTING(F_SOUNDSETTING|F_NO_WRAP, compressor_knee,
LANG_COMPRESSOR_KNEE, 1, "compressor knee",
"hard knee,soft knee", compressor_set, 2,
ID2P(LANG_COMPRESSOR_HARD_KNEE), ID2P(LANG_COMPRESSOR_SOFT_KNEE)),
INT_SETTING_NOWRAP(F_SOUNDSETTING, compressor_release_time,
LANG_COMPRESSOR_RELEASE, 100,
"compressor release time", UNIT_MS, 20, 200,
10, NULL, NULL, compressor_set),
#endif
#ifdef HAVE_WM8758
SOUND_SETTING(F_NO_WRAP, bass_cutoff, LANG_BASS_CUTOFF,