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authorJeffrey Goode <jeffg7@gmail.com>2009-08-18 03:24:45 +0000
committerJeffrey Goode <jeffg7@gmail.com>2009-08-18 03:24:45 +0000
commit2b7ef6b92880249e64639768012266e65f5d14cc (patch)
tree119d27b90177426b8c01da7d3f5a4e1d884a01ee
parentf451108fa857fe16f35fa3beb5278e36fed74c67 (diff)
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FS#10199: Adds limiter DSP function
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@22394 a1c6a512-1295-4272-9138-f99709370657
-rw-r--r--apps/dsp.c464
-rw-r--r--apps/dsp.h4
-rw-r--r--apps/lang/english.lang17
-rw-r--r--apps/menus/sound_menu.c3
-rw-r--r--apps/pcmbuf.c24
-rw-r--r--apps/pcmbuf.h1
-rw-r--r--apps/plugin.c2
-rw-r--r--apps/plugin.h1
-rw-r--r--apps/plugins/test_codec.c24
-rw-r--r--apps/settings.c1
-rw-r--r--apps/settings.h5
-rw-r--r--apps/settings_list.c6
12 files changed, 546 insertions, 6 deletions
diff --git a/apps/dsp.c b/apps/dsp.c
index 61fc027e4c..c8c8ddee75 100644
--- a/apps/dsp.c
+++ b/apps/dsp.c
@@ -35,6 +35,11 @@
#include "buffer.h"
#include "fixedpoint.h"
#include "fracmul.h"
+#include "pcmbuf.h"
+
+/* Define LOGF_ENABLE to enable logf output in this file */
+/*#define LOGF_ENABLE*/
+#include "logf.h"
/* 16-bit samples are scaled based on these constants. The shift should be
* no more than 15.
@@ -149,7 +154,8 @@ 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[]);
/*
***************************************************************************/
@@ -165,6 +171,7 @@ 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;
@@ -180,6 +187,7 @@ 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;
};
/* General DSP config */
@@ -219,6 +227,55 @@ static long album_peak;
static long replaygain;
static bool crossfeed_enabled;
+/* limiter */
+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[LIMITER_BUFFER_SIZE] IBSS_ATTR;
+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};
+
#define AUDIO_DSP (dsp_conf[CODEC_IDX_AUDIO])
#define VOICE_DSP (dsp_conf[CODEC_IDX_VOICE])
@@ -869,6 +926,7 @@ static void dsp_apply_gain(int count, struct dsp_data *data, int32_t *buf[])
/* Combine all gains to a global gain. */
static void set_gain(struct dsp_config *dsp)
{
+ /* gains are in S7.24 format */
dsp->data.gain = DEFAULT_GAIN;
/* Replay gain not relevant to voice */
@@ -879,8 +937,14 @@ static void set_gain(struct dsp_config *dsp)
if (dsp->eq_process && eq_precut)
{
- dsp->data.gain =
- (long) (((int64_t) dsp->data.gain * eq_precut) >> 24);
+ 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
@@ -898,7 +962,7 @@ static void set_gain(struct dsp_config *dsp)
}
else
{
- dsp->data.gain >>= 1;
+ dsp->data.gain >>= 1; /* convert gain to S8.23 format */
}
dsp->apply_gain = dsp->data.gain != 0 ? dsp_apply_gain : NULL;
@@ -1207,7 +1271,7 @@ int dsp_process(struct dsp_config *dsp, char *dst, const char *src[], int count)
if (dsp->tdspeed_active)
samples = tdspeed_doit(tmp, samples);
-
+
int chunk_offset = 0;
while (samples > 0)
{
@@ -1239,6 +1303,9 @@ 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);
dsp->output_samples(chunk, &dsp->data, (const int32_t **)t2, (int16_t *)dst);
@@ -1286,6 +1353,15 @@ 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;
}
@@ -1295,6 +1371,13 @@ 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)
@@ -1412,6 +1495,7 @@ 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);
break;
case DSP_FLUSH:
@@ -1420,6 +1504,7 @@ 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);
break;
case DSP_SET_TRACK_GAIN:
@@ -1498,3 +1583,372 @@ void dsp_set_replaygain(void)
replaygain = gain;
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)
+ {
+ 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;
+ }
+}
+
+/** 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 */
+ {
+ 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;
+ }
+ }
+ }
+ reset_limiter_buffer(&AUDIO_DSP);
+ }
+ 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]);
+ 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;
+ 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)
+ {
+ /* 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.
+ */
+static int limiter_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);
+
+ 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
+ * 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)
+ {
+ release_peak = out_buf_peak[i];
+ }
+ /* keep sloping */
+ else
+ {
+ release_peak--;
+ out_buf_peak[i] = release_peak;
+ }
+ }
+ /* Implement the limiter: adjust gain of the outbound samples by the gain
+ * amounts in the gain steps array corresponding to the peak values. */
+ for (ch = 0; ch < AUDIO_DSP.data.num_channels; ch++)
+ {
+ int32_t *d = buf[ch];
+ for (i = 0; i < count; i++)
+ {
+ if (out_buf_peak[i] > 0)
+ {
+ 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);
+ d[i] = FRACMUL_SHL(d[i], gain, 3);
+ }
+ }
+ }
+ return count;
+}
diff --git a/apps/dsp.h b/apps/dsp.h
index 58a5edb5e2..b2d8493445 100644
--- a/apps/dsp.h
+++ b/apps/dsp.h
@@ -26,6 +26,8 @@
#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,
@@ -80,5 +82,7 @@ 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);
#endif
diff --git a/apps/lang/english.lang b/apps/lang/english.lang
index 23ab26aba6..53de28128f 100644
--- a/apps/lang/english.lang
+++ b/apps/lang/english.lang
@@ -12699,3 +12699,20 @@
lcd_bitmap: "Scroll bar position"
</voice>
</phrase>
+<phrase>
+ id: LANG_LIMITER
+ desc: in sound settings
+ user: core
+ <source>
+ *: none
+ swcodec: "Limiter Preamp"
+ </source>
+ <dest>
+ *: none
+ swcodec: "Limiter Preamp"
+ </dest>
+ <voice>
+ *: none
+ swcodec: "Limiter Preamp"
+ </voice>
+</phrase>
diff --git a/apps/menus/sound_menu.c b/apps/menus/sound_menu.c
index 1723d9779e..678d495d07 100644
--- a/apps/menus/sound_menu.c
+++ b/apps/menus/sound_menu.c
@@ -105,6 +105,8 @@ 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);
#endif
#if (CONFIG_CODEC == MAS3587F) || (CONFIG_CODEC == MAS3539F)
@@ -138,6 +140,7 @@ 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
#endif
#if (CONFIG_CODEC == MAS3587F) || (CONFIG_CODEC == MAS3539F)
,&loudness,&avc,&superbass,&mdb_enable,&mdb_strength
diff --git a/apps/pcmbuf.c b/apps/pcmbuf.c
index 319e3e8044..66a4ed4128 100644
--- a/apps/pcmbuf.c
+++ b/apps/pcmbuf.c
@@ -1170,6 +1170,30 @@ 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);
+}
diff --git a/apps/pcmbuf.h b/apps/pcmbuf.h
index 4cb1c1acb7..12a41a253c 100644
--- a/apps/pcmbuf.h
+++ b/apps/pcmbuf.h
@@ -76,5 +76,6 @@ 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
diff --git a/apps/plugin.c b/apps/plugin.c
index e9acc97824..674740cf83 100644
--- a/apps/plugin.c
+++ b/apps/plugin.c
@@ -668,7 +668,7 @@ static const struct plugin_api rockbox_api = {
appsversion,
/* new stuff at the end, sort into place next time
the API gets incompatible */
-
+ dsp_flush_limiter_buffer,
};
int plugin_load(const char* plugin, const void* parameter)
diff --git a/apps/plugin.h b/apps/plugin.h
index 9616628e90..d47cf031ff 100644
--- a/apps/plugin.h
+++ b/apps/plugin.h
@@ -836,6 +836,7 @@ struct plugin_api {
const char *appsversion;
/* new stuff at the end, sort into place next time
the API gets incompatible */
+ int (*dsp_flush_limiter_buffer)(char *dest);
};
/* plugin header */
diff --git a/apps/plugins/test_codec.c b/apps/plugins/test_codec.c
index bae4571067..22516ba373 100644
--- a/apps/plugins/test_codec.c
+++ b/apps/plugins/test_codec.c
@@ -810,10 +810,34 @@ 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();
diff --git a/apps/settings.c b/apps/settings.c
index 48571db20e..b76468280e 100644
--- a/apps/settings.c
+++ b/apps/settings.c
@@ -950,6 +950,7 @@ 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);
#endif
#ifdef HAVE_SPDIF_POWER
diff --git a/apps/settings.h b/apps/settings.h
index 2fc07078ce..c33b715b95 100644
--- a/apps/settings.h
+++ b/apps/settings.h
@@ -771,6 +771,11 @@ struct user_settings
int scrollbar_width;
bool scrollbar_position; /* true=left false=right */
#endif
+
+#if CONFIG_CODEC == SWCODEC
+ int limiter_level;
+#endif
+
};
/** global variables **/
diff --git a/apps/settings_list.c b/apps/settings_list.c
index 86972a132d..9ea95573cf 100644
--- a/apps/settings_list.c
+++ b/apps/settings_list.c
@@ -1246,6 +1246,12 @@ const struct settings_list settings[] = {
/* timestretch */
OFFON_SETTING(F_SOUNDSETTING, timestretch_enabled, LANG_TIMESTRETCH, false,
"timestretch enabled", dsp_timestretch_enable),
+
+ /* limiter */
+ INT_SETTING_NOWRAP(F_SOUNDSETTING, limiter_level,
+ LANG_LIMITER, 0,
+ "limiter level", UNIT_DB, 0, MAX_LIMITER_GAIN,
+ 5, db_format, get_dec_talkid, dsp_set_limiter),
#endif
#ifdef HAVE_WM8758
SOUND_SETTING(F_NO_WRAP, bass_cutoff, LANG_BASS_CUTOFF,