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Compressor: save lots of RAM, bug fix to work with internally clipped samples

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@23268 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Jeffrey Goode 2009-10-19 19:40:12 +00:00
parent 334e03e55b
commit c8fa54ecfa
1 changed files with 125 additions and 85 deletions
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210
apps/dsp.c
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@ -163,8 +163,6 @@ typedef void (*channels_process_fn_type)(int count, int32_t *buf[]);
/* DSP local channel processing in place */
typedef void (*channels_process_dsp_fn_type)(int count, struct dsp_data *data,
int32_t *buf[]);
/* DSP processes that return a value */
typedef int (*return_fn_type)(int count, int32_t *buf[]);
/*
***************************************************************************/
@ -195,7 +193,7 @@ struct dsp_config
channels_process_fn_type apply_crossfeed;
channels_process_fn_type eq_process;
channels_process_fn_type channels_process;
return_fn_type compressor_process;
channels_process_fn_type compressor_process;
};
/* General DSP config */
@ -262,16 +260,13 @@ static int32_t *resample_buf;
#define RESAMPLE_BUF_RIGHT_CHANNEL (sample_buf_count/2 * RESAMPLE_RATIO)
/* 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[]);
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[]);
/* Clip sample to signed 16 bit range */
@ -1269,7 +1264,7 @@ int dsp_process(struct dsp_config *dsp, char *dst, const char *src[], int count)
dsp->channels_process(chunk, t2);
if (dsp->compressor_process)
chunk = dsp->compressor_process(chunk, t2);
dsp->compressor_process(chunk, t2);
dsp->output_samples(chunk, &dsp->data, (const int32_t **)t2, (int16_t *)dst);
@ -1444,7 +1439,7 @@ intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value)
resampler_new_delta(dsp);
tdspeed_setup(dsp);
if (dsp == &AUDIO_DSP)
release_gain = (1 << 24);
release_gain = UNITY;
break;
case DSP_FLUSH:
@ -1454,7 +1449,7 @@ intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value)
dither_init(dsp);
tdspeed_setup(dsp);
if (dsp == &AUDIO_DSP)
release_gain = (1 << 24);
release_gain = UNITY;
break;
case DSP_SET_TRACK_GAIN:
@ -1614,13 +1609,15 @@ void dsp_set_compressor(int c_threshold, int c_ratio, int c_gain,
{
int32_t db; /* S15.16 format */
int32_t offset; /* S15.16 format */
} db_curve[4];
} 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 */
[3] = 0 db input
[4] = ~+12db input (2 bits clipping overhead) */
db_curve[1].db = c_menu.threshold << 16;
if (c_menu.soft_knee)
{
@ -1644,37 +1641,61 @@ void dsp_set_compressor(int c_threshold, int c_ratio, int c_gain,
db_curve[2].db = c_menu.threshold << 16;
db_curve[2].offset = 0;
}
/* 0db input is also max offset point (most compression) */
/* Calculate 0db and ~+12db offsets */
db_curve[4].db = 0xC0A8C; /* db of 2 bits clipping */
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);
db_curve[4].offset = (int32_t)((long long)-db_curve[4].db
* (c_menu.ratio - 1) / c_menu.ratio) + db_curve[3].offset;
}
else
{
/* offset = threshold for hard limit */
db_curve[3].offset = (c_menu.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] = (1 << 24);
/** 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 (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] = (1 << 24);
comp_curve[i] = UNITY;
/* if soft knee and below top of knee, interpolate along soft knee slope */
/* 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),
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[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(SIMULATOR) && defined(LOGF_ENABLE)
logf("\n *** Compression Offsets ***");
/* some settings for display only, not used in calculations */
@ -1682,31 +1703,33 @@ void dsp_set_compressor(int c_threshold, int c_ratio, int c_gain,
db_curve[1].offset = 0;
db_curve[3].db = 0;
for (i = 0; i <= 3; i++)
for (i = 0; i <= 4; i++)
{
logf("Curve[%d]: db: % .1f\toffset: % .4f", i, (float)db_curve[i].db / (1 << 16),
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 (i = 1; i <= 64; i++)
for (i = 1; i <= 65; i++)
{
debugf("%02d: %.6f ", i, (float)comp_curve[i] / (1 << 24));
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 */
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));
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 / c_menu.release;
logf("Release slope:\t%.6f", (float)comp_rel_slope / (1 << 24));
logf("Release slope:\t%.6f", (float)comp_rel_slope / UNITY);
release_gain = (1 << 24);
release_gain = UNITY;
}
/* enable/disable the compressor */
@ -1719,83 +1742,100 @@ void dsp_set_compressor(int c_threshold, int c_ratio, int c_gain,
*/
static inline int32_t get_compression_gain(int32_t sample)
{
const int frac_bits = AUDIO_DSP.frac_bits;
const int frac_bits_offset = AUDIO_DSP.frac_bits - 15;
/* 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);
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;
/* index is 6 MSB, rem is 16 LSB */
int index = sample >> 16;
int rem = (sample & 0xFFFF) << 8;
/* 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])));
}
/* interpolate from the compression curve */
return comp_curve[index] + (int32_t)FRACMUL_SHL((comp_curve[index + 1]
- comp_curve[index]), rem, 7);
/* sample is too clipped, return invalid value */
return -1;
}
/** COMPRESSOR PROCESS
* Changes the gain of the samples according to the compressor curve
*/
static int compressor_process(int count, int32_t *buf[])
static void 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 *in_buf[2] = {buf[0], buf[1]};
int32_t sample_gain, /* S7.24 format */
this_gain; /* S7.24 format */
int i, ch;
/* Step forward through the output buffer, and modify the offset values
* to establish a smooth, slow release slope.*/
for (i = 0; i < count; i++)
while (count-- > 0)
{
sample_gain = fp_one;
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++)
{
this_gain = get_compression_gain(buf[ch][i]);
int32_t this_gain = get_compression_gain(*in_buf[ch]);
if (this_gain < sample_gain)
sample_gain = this_gain;
}
/* if no release slope, only apply makeup gain */
if ((sample_gain == fp_one) && (release_gain == fp_one))
gain_buffer[i] = comp_makeup_gain;
else
/* perform release slope; skip if no compression and no release slope */
if ((sample_gain != UNITY) || (release_gain != UNITY))
{
/* if larger offset, start release slope */
if (sample_gain <= release_gain)
release_gain = sample_gain;
else /* keep sloping */
/* if larger offset than previous slope, start new release slope */
if ((sample_gain <= release_gain) && (sample_gain > 0))
{
if (release_gain < (fp_one - comp_rel_slope))
release_gain += comp_rel_slope;
else
release_gain = fp_one;
release_gain = sample_gain;
}
/* 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);
/* keep sloping towards unity gain (and ignore invalid value) */
{
release_gain += comp_rel_slope;
if (release_gain > UNITY)
{
release_gain = UNITY;
}
}
}
}
/* Implement the compressor: apply those gain factors to the output
* buffer samples */
for (i = 0; i < count; i++)
{
if (gain_buffer[i] != fp_one)
/* 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++)
buf[ch][i] = FRACMUL_SHL(buf[ch][i], gain_buffer[i], 7);
{
*in_buf[ch] = FRACMUL_SHL(total_gain, *in_buf[ch], 7);
}
}
in_buf[0]++;
in_buf[1]++;
}
return count;
}