Index: apps/pcmbuf.c
===================================================================
--- apps/pcmbuf.c (revision 21906)
+++ apps/pcmbuf.c (working copy)
@@ -446,7 +446,7 @@
/* Buffer has to be at least 2s long. */
seconds += 2;
#endif
- logf("pcmbuf len: %ld", seconds);
+ logf("pcmbuf len: %ld", (long)seconds);
return seconds * (NATIVE_FREQUENCY*4); /* 2 channels + 2 bytes/sample */
}
@@ -1155,8 +1155,28 @@
return crossfade_enabled;
}
+/** PLAY LAST REMAINING SAMPLES AT END OF PLAYBACK
+ * Empty the attack buffer and commit any remaining
+ * samples in the PCM buffer to playback. */
void pcmbuf_play_remainder(void)
{
+ char *dest;
+ int out_count = ATTACK_BUFFER_SIZE;
+
+ /* create room at the end of the PCM buffer for the
+ samples held back in the attack buffer */
+ while ((dest = pcmbuf_request_buffer(&out_count)) == NULL)
+ {
+ cancel_cpu_boost();
+ sleep(1);
+ }
+
+ /* flush the attack buffer into the PCM buffer */
+ out_count = dsp_flush_attack_buffer(dest);
+ if (out_count > 0)
+ pcmbuf_write_complete(out_count);
+
+ /* commit remaining samples to playback */
if (audiobuffer_fillpos)
pcmbuf_flush_fillpos();
}
Index: apps/settings.c
===================================================================
--- apps/settings.c (revision 21906)
+++ apps/settings.c (working copy)
@@ -939,7 +939,10 @@
}
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
Index: apps/plugins/mpegplayer/audio_thread.c
===================================================================
--- apps/plugins/mpegplayer/audio_thread.c (revision 21906)
+++ apps/plugins/mpegplayer/audio_thread.c (working copy)
@@ -482,7 +482,7 @@
td.dsp = (struct dsp_config *)rb->dsp_configure(NULL, DSP_MYDSP,
CODEC_IDX_AUDIO);
rb->sound_set_pitch(PITCH_SPEED_100);
- rb->dsp_configure(td.dsp, DSP_RESET, 0);
+ rb->dsp_configure(td.dsp, DSP_RESET_NO_ATTACK_BUFFER, 0);
rb->dsp_configure(td.dsp, DSP_SET_SAMPLE_DEPTH, MAD_F_FRACBITS);
goto message_wait;
Index: apps/lang/english.lang
===================================================================
--- apps/lang/english.lang (revision 21906)
+++ apps/lang/english.lang (working copy)
@@ -12655,3 +12655,20 @@
pitchscreen: "Rate"
+
+ id: LANG_LIMITER
+ desc: in sound settings
+ user: core
+
+ *: none
+ swcodec: "Limiter Preamp"
+
+
+ *: none
+ swcodec: "Limiter Preamp"
+
+
+ *: none
+ swcodec: "Limiter Preamp"
+
+
Index: apps/settings.h
===================================================================
--- apps/settings.h (revision 21906)
+++ apps/settings.h (working copy)
@@ -746,6 +746,11 @@
bool pitch_mode_timestretch;
#endif
#endif
+
+#if CONFIG_CODEC == SWCODEC
+ int limiter_level;
+#endif
+
/* If values are just added to the end, no need to bump plugin API
version. */
/* new stuff to be added at the end */
Index: apps/menus/sound_menu.c
===================================================================
--- apps/menus/sound_menu.c (revision 21906)
+++ apps/menus/sound_menu.c (working copy)
@@ -105,6 +105,9 @@
&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 +141,7 @@
#if CONFIG_CODEC == SWCODEC
,&crossfeed_menu, &equalizer_menu, &dithering_enabled
,×tretch_enabled
+ ,&limiter_level
#endif
#if (CONFIG_CODEC == MAS3587F) || (CONFIG_CODEC == MAS3539F)
,&loudness,&avc,&superbass,&mdb_enable,&mdb_strength
Index: apps/dsp.c
===================================================================
--- apps/dsp.c (revision 21906)
+++ apps/dsp.c (working copy)
@@ -36,6 +36,10 @@
#include "fixedpoint.h"
#include "fracmul.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.
*/
@@ -149,8 +153,8 @@
/* DSP local channel processing in place */
typedef void (*channels_process_dsp_fn_type)(int count, struct dsp_data *data,
int32_t *buf[]);
+typedef int (*attack_fn_type)(int count, int32_t *ext_buf[]);
-
/*
***************************************************************************/
@@ -165,6 +169,7 @@
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 +185,8 @@
channels_process_fn_type apply_crossfeed;
channels_process_fn_type eq_process;
channels_process_fn_type channels_process;
+ attack_fn_type set_attack_buffer;
+ channels_process_fn_type limiter_process;
};
/* General DSP config */
@@ -219,6 +226,49 @@
static long replaygain;
static bool crossfeed_enabled;
+
+/* attack buffer */
+static int count_adjust;
+static bool attack_buffer_full;
+static bool attack_buffer_emptying;
+static int32_t attack_buffer[2][ATTACK_BUFFER_SIZE] IBSS_ATTR;
+static int32_t *start_attack[2], *end_attack[2];
+static uint16_t attack_peak[ATTACK_BUFFER_SIZE] IBSS_ATTR;
+static uint16_t *start_peak, *end_peak;
+static uint16_t buf_peak[ATTACK_BUFFER_SIZE] IBSS_ATTR;
+
+static void clear_attack_buffer(void);
+static int set_attack_buffer(int count, int32_t *buf[]);
+static int attack_buffer_count(bool empty_count);
+
+/* limiter */
+static uint16_t release_peak;
+
+static void limiter_process(int count, int32_t *buf[]);
+static uint16_t get_peak_value(int32_t sample);
+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};
+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])
@@ -245,7 +295,6 @@
#define RESAMPLE_BUF_LEFT_CHANNEL 0
#define RESAMPLE_BUF_RIGHT_CHANNEL (sample_buf_count/2 * RESAMPLE_RATIO)
-
/* Clip sample to signed 16 bit range */
static inline int32_t clip_sample_16(int32_t sample)
{
@@ -477,13 +526,12 @@
const int scale = data->output_scale;
const int dc_bias = 1 << (scale - 1);
- do
+ while (count-- > 0)
{
int32_t lr = clip_sample_16((*s0++ + dc_bias) >> scale);
*dst++ = lr;
*dst++ = lr;
}
- while (--count > 0);
}
#endif /* DSP_HAVE_ASM_SAMPLE_OUTPUT_MONO */
@@ -497,12 +545,11 @@
const int scale = data->output_scale;
const int dc_bias = 1 << (scale - 1);
- do
+ while (count-- > 0)
{
*dst++ = clip_sample_16((*s0++ + dc_bias) >> scale);
*dst++ = clip_sample_16((*s1++ + dc_bias) >> scale);
}
- while (--count > 0);
}
#endif /* DSP_HAVE_ASM_SAMPLE_OUTPUT_STEREO */
@@ -876,6 +923,7 @@
/* 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 */
@@ -886,17 +934,23 @@
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 */
+ if ((dsp->limiter_preamp) && (dsp->frac_bits <= 29))
+ {
+ dsp->data.gain = fp_mul(dsp->data.gain, dsp->limiter_preamp, 24);
+ }
+
if (dsp->data.gain == DEFAULT_GAIN)
{
dsp->data.gain = 0;
}
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;
@@ -1194,18 +1248,27 @@
/* Testing function pointers for NULL is preferred since the pointer
will be preloaded to be used for the call if not. */
- while (count > 0)
+ while ((count > 0) || (attack_buffer_emptying))
{
int samples = MIN(sample_buf_count/2, count);
count -= samples;
- dsp->input_samples(samples, src, tmp);
+ if (attack_buffer_emptying)
+ {
+ /* point to flush buffer */
+ tmp[0] = (int32_t *)src[0];
+ tmp[1] = (int32_t *)src[1];
+ }
+ else
+ {
+ dsp->input_samples(samples, src, tmp);
- if (dsp->tdspeed_active)
- samples = tdspeed_doit(tmp, samples);
-
+ if (dsp->tdspeed_active)
+ samples = tdspeed_doit(tmp, samples);
+ }
+
int chunk_offset = 0;
- while (samples > 0)
+ while ((samples > 0) || (attack_buffer_emptying))
{
int32_t *t2[2];
t2[0] = tmp[0]+chunk_offset;
@@ -1215,27 +1278,39 @@
chunk_offset += chunk;
samples -= chunk;
- if (dsp->apply_gain)
- dsp->apply_gain(chunk, &dsp->data, t2);
+ if (!attack_buffer_emptying) /* don't process when chunk = 0 */
+ {
+ if (dsp->apply_gain)
+ dsp->apply_gain(chunk, &dsp->data, t2);
- if (dsp->resample && (chunk = resample(dsp, chunk, t2)) <= 0)
- break; /* I'm pretty sure we're downsampling here */
+ if (dsp->resample && (chunk = resample(dsp, chunk, t2)) <= 0)
+ break; /* I'm pretty sure we're downsampling here */
- if (dsp->apply_crossfeed)
- dsp->apply_crossfeed(chunk, t2);
+ if (dsp->apply_crossfeed)
+ dsp->apply_crossfeed(chunk, t2);
- if (dsp->eq_process)
- dsp->eq_process(chunk, t2);
+ if (dsp->eq_process)
+ dsp->eq_process(chunk, t2);
#ifdef HAVE_SW_TONE_CONTROLS
- if ((bass | treble) != 0)
- eq_filter(t2, &dsp->tone_filter, chunk,
- dsp->data.num_channels, FILTER_BISHELF_SHIFT);
+ if ((bass | treble) != 0)
+ eq_filter(t2, &dsp->tone_filter, chunk,
+ dsp->data.num_channels, FILTER_BISHELF_SHIFT);
#endif
- if (dsp->channels_process)
- dsp->channels_process(chunk, t2);
+ if (dsp->channels_process)
+ dsp->channels_process(chunk, t2);
+ } /* !attack_buffer_emptying */
+
+ /* operate the attack buffer */
+ if (dsp->set_attack_buffer)
+ chunk = set_attack_buffer(chunk, t2);
+ /* put all processes that refer to the attack buffer after this line */
+
+ if (dsp->limiter_process)
+ dsp->limiter_process(chunk, t2);
+
dsp->output_samples(chunk, &dsp->data, (const int32_t **)t2, (int16_t *)dst);
written += chunk;
@@ -1282,6 +1357,16 @@
*/
if (count > RESAMPLE_BUF_RIGHT_CHANNEL)
count = RESAMPLE_BUF_RIGHT_CHANNEL;
+
+ /* If the attack buffer is filling, some or all samples will
+ * be captured by it, so expect fewer samples coming out.
+ */
+ if (dsp->set_attack_buffer && !attack_buffer_full)
+ {
+ int empty_space = attack_buffer_count(true);
+ count_adjust = MIN(empty_space, count);
+ count -= count_adjust;
+ }
return count;
}
@@ -1291,6 +1376,13 @@
*/
int dsp_input_count(struct dsp_config *dsp, int count)
{
+ /* If the attack buffer is filling, the output count was
+ * adjusted downward. This adjusts it back so that input
+ * count is not affected.
+ */
+ if (dsp->set_attack_buffer && !attack_buffer_full)
+ count += count_adjust;
+
/* count is now the number of resampled input samples. Convert to
original input samples. */
if (dsp->resample)
@@ -1385,7 +1477,8 @@
tdspeed_setup(dsp);
break;
- case DSP_RESET:
+ case DSP_RESET: /* fall through! */
+ case DSP_RESET_NO_ATTACK_BUFFER:
dsp->stereo_mode = STEREO_NONINTERLEAVED;
dsp->data.num_channels = 2;
dsp->sample_depth = NATIVE_DEPTH;
@@ -1408,6 +1501,11 @@
dsp_update_functions(dsp);
resampler_new_delta(dsp);
tdspeed_setup(dsp);
+ dsp->set_attack_buffer =
+ ((dsp == &AUDIO_DSP) && (setting != DSP_RESET_NO_ATTACK_BUFFER))
+ ? set_attack_buffer : NULL;
+ clear_attack_buffer();
+ release_peak = 0;
break;
case DSP_FLUSH:
@@ -1416,6 +1514,8 @@
resampler_new_delta(dsp);
dither_init(dsp);
tdspeed_setup(dsp);
+ clear_attack_buffer();
+ release_peak = 0;
break;
case DSP_SET_TRACK_GAIN:
@@ -1490,7 +1590,395 @@
}
}
- /* Store in S8.23 format to simplify calculations. */
+ /* Store in S7.24 format to simplify calculations. */
replaygain = gain;
set_gain(&AUDIO_DSP);
}
+
+
+/** CLEAR THE ATTACK BUFFER
+ * Forces the attack buffer to the initial state and discard
+ * any samples held there. */
+static void clear_attack_buffer(void)
+{
+ if (AUDIO_DSP.set_attack_buffer == NULL) return;
+ int i;
+ logf("clear_attack_buffer");
+ for (i = 0; i < 2; i++)
+ {
+ start_attack[i] = end_attack[i] = attack_buffer[i];
+ start_peak = end_peak = attack_peak;
+ }
+ attack_buffer_full = false;
+ attack_buffer_emptying = false;
+}
+
+
+/** OPERATE THE ATTACK BUFFER
+ * Handles all samples entering or exiting the attack buffer.
+ * This should only be called from dsp_process, and dsp_process should
+ * only be called after dsp_output_count and dsp_input_count. */
+static int set_attack_buffer(int count, int32_t *buf[])
+{
+ int32_t *in_buf[2], *out_buf[2], temp, samp0 = 0;
+ int empty_space,
+ i,
+ out_count = 0;
+ uint16_t *buf_peak_index = buf_peak;
+ const long clip_max = AUDIO_DSP.data.clip_max;
+
+ in_buf[0] = buf[0];
+ in_buf[1] = buf[1];
+ out_buf[0] = buf[0];
+ out_buf[1] = buf[1];
+
+ if (attack_buffer_emptying)
+ /** EMPTY THE BUFFER
+ * since the empty flag has been set, assume no inbound samples and
+ return all samples in the attack buffer to the outbound buffer */
+ {
+ count = attack_buffer_count(false);
+ out_count = count;
+ logf("Emptying attack buffer: %d", count);
+ while (count-- > 0)
+ {
+ for (i = 0; i < 2; i++)
+ {
+ /* move samples in attack buffer to output buffer */
+ *out_buf[i]++ = *start_attack[i]++;
+ if (start_attack[i] == &attack_buffer[i][ATTACK_BUFFER_SIZE])
+ start_attack[i] = attack_buffer[i];
+ /* move attack buffer peak values to output peak values */
+ if (i == 0)
+ {
+ *buf_peak_index++ = *start_peak++;
+ if (start_peak == &attack_peak[ATTACK_BUFFER_SIZE])
+ start_peak = attack_peak;
+ }
+ }
+ }
+ clear_attack_buffer();
+ }
+ else /* attack buffer NOT emptying */
+ {
+ empty_space = attack_buffer_count(true);
+
+ 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 attack buffer: %d", empty_space);
+ do
+ {
+ for (i = 0; i < 2; i++)
+ {
+ /* put inbound samples in the attack buffer */
+ temp = *in_buf[i]++;
+ *end_attack[i]++ = temp;
+ if (end_attack[i] == &attack_buffer[i][ATTACK_BUFFER_SIZE])
+ end_attack[i] = attack_buffer[i];
+ if (temp < 0) /* make positive for comparison */
+ temp = -temp - 1;
+ if (temp <= clip_max)
+ temp = 0; /* disregard if not clipped */
+ if (i == 0)
+ {
+ samp0 = temp;
+ }
+ else
+ {
+ /* assign peak value for each inbound sample pair */
+ *end_peak++ = ((samp0 > 0) || (temp > 0)) ?
+ get_peak_value(MAX(samp0, temp)) : 0;
+ if (end_peak == &attack_peak[ATTACK_BUFFER_SIZE])
+ end_peak = attack_peak;
+ }
+ }
+ count--;
+ }
+ while (--empty_space > 0);
+ /* after buffer fills, the remaining inbound samples are cycled */
+ }
+
+ attack_buffer_full = (end_attack[0] == start_attack[0]);
+
+ if (count > 0)
+ /** CYCLE BUFFER
+ * return buffered samples and backfill attack buffer with new ones */
+ {
+ /* buffer should always be full here */
+#ifdef SIMULATOR
+ if (!attack_buffer_full)
+ DEBUGF("Attack buffer cycle error: %d / %d\n",
+ attack_buffer_count(false), ATTACK_BUFFER_SIZE);
+#endif
+ out_count = count;
+ do
+ {
+ for (i = 0; i < 2; i++)
+ {
+ /* copy incoming sample */
+ temp = *in_buf[i]++;
+ /* put attack buffer sample into outbound buffer */
+ *out_buf[i]++ = *start_attack[i]++;
+ /* put incoming sample on the end of the attack buffer */
+ *end_attack[i]++ = temp;
+ /* ring buffer pointer wrap */
+ if (start_attack[i] == &attack_buffer[i][ATTACK_BUFFER_SIZE])
+ start_attack[i] = attack_buffer[i];
+ if (end_attack[i] == &attack_buffer[i][ATTACK_BUFFER_SIZE])
+ end_attack[i] = attack_buffer[i];
+ if (temp < 0) /* make positive for comparison */
+ temp = -temp - 1;
+ if (temp <= clip_max)
+ temp = 0; /* disregard if not clipped */
+ if (i == 0)
+ {
+ samp0 = temp;
+ /* assign outgoing sample its associated peak value */
+ *buf_peak_index++ = *start_peak++;
+ if (start_peak == &attack_peak[ATTACK_BUFFER_SIZE])
+ start_peak = attack_peak;
+ }
+ else
+ {
+ /* assign peak value for each inbound sample pair */
+ *end_peak++ = ((samp0 > 0) || (temp > 0)) ?
+ get_peak_value(MAX(samp0, temp)) : 0;
+ if (end_peak == &attack_peak[ATTACK_BUFFER_SIZE])
+ end_peak = attack_peak;
+ }
+ }
+ }
+ while (--count > 0);
+ }
+ }
+
+ return out_count;
+}
+
+/** RETURN ATTACK BUFFER COUNT
+ * If argument is true, returns empty space remaining,
+ * otherwise, returns number of samples in the buffer */
+static int attack_buffer_count(bool empty_count)
+{
+ int count;
+ if (attack_buffer_full)
+ count = ATTACK_BUFFER_SIZE;
+ else if (end_attack[0] >= start_attack[0])
+ count = (end_attack[0] - start_attack[0]);
+ else
+ count = (end_attack[0] - start_attack[0]) + ATTACK_BUFFER_SIZE;
+ return empty_count ? (ATTACK_BUFFER_SIZE - count) : count;
+}
+
+
+/** FLUSH THE ATTACK BUFFER
+ * Only called by the pcmbuf_play_remainder routine in pcmbuf.c
+ * Empties the attack buffer into the buffer pointed to by the argument
+ * and returns the number of samples in that buffer */
+int dsp_flush_attack_buffer(char *dest)
+{
+ logf("dsp_flush_attack_buffer");
+ if ((AUDIO_DSP.set_attack_buffer == NULL) ||
+ (attack_buffer_count(false) <= 0))
+ return 0;
+
+ int32_t flush_buf[2][ATTACK_BUFFER_SIZE];
+ const char *src[2];
+ src[0] = (char *)flush_buf[0];
+ src[1] = (char *)flush_buf[1];
+
+ attack_buffer_emptying = true;
+
+ return dsp_process(&AUDIO_DSP, dest, src, 0);
+}
+
+
+/** 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)
+{
+ /* 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 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]));
+
+#if 0
+ /* old working code - keeping this for now in case new stuff breaks... */
+ for (i = 1; i <= 48; i++)
+ {
+ if (sample < clip_steps[i])
+ {
+ /* interpolate linearly between steps (less accurate but faster) */
+ return ((i-1) * 90) + (((sample - clip_steps[i-1]) * 90) /
+ (clip_steps[i] - clip_steps[i-1]));
+ }
+ }
+ return 48*90;
+#endif
+}
+
+
+void dsp_set_limiter(int limiter_level)
+{
+ if (limiter_level > 0)
+ {
+ /* enable limiter process */
+ AUDIO_DSP.limiter_process = limiter_process;
+ /* limiter preamp is a gain factor in S7.24 format */
+ AUDIO_DSP.limiter_preamp =
+ fp_factor((((long)limiter_level << 24) / 10), 24);
+ }
+ else
+ {
+ /* disable limiter process*/
+ AUDIO_DSP.limiter_process = NULL;
+ AUDIO_DSP.limiter_preamp = 0;
+ }
+
+ set_gain(&AUDIO_DSP);
+
+ logf("Limiter enable: %s\tLimiter amp: %.8f",
+ AUDIO_DSP.limiter_process ? "Yes" : "No",
+ (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 void limiter_process(int count, int32_t *buf[])
+{
+ /* Limiter process passes through if attack buffer isn't active, or the
+ * sample depth is too large for safe pre-amping, or there is nothing
+ * to process. */
+ if ((AUDIO_DSP.set_attack_buffer == NULL) ||
+ (AUDIO_DSP.frac_bits > 29) || (count <= 0)) return;
+
+ /* 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 int attack_slope = 15; /* 15:1 ratio between attack and release */
+ const int attack_count = attack_buffer_count(false);
+
+ int i, ch;
+ uint16_t attack_peak_i,
+ max_peak = 0,
+ gain_peak,
+ gain_rem;
+ long gain;
+
+ /* step through attack buffer in reverse order, in order to find the
+ * appropriate max_peak for modifying the output buffer */
+ for (i = attack_count - 1; i >= 0; i--)
+ {
+ attack_peak_i = attack_peak[(start_peak - attack_peak + i) %
+ ATTACK_BUFFER_SIZE];
+ /* if no attack slope, nothing to do */
+ if ((attack_peak_i == 0) && (max_peak == 0)) continue;
+ /* if new peak, start attack slope */
+ if (attack_peak_i >= max_peak)
+ {
+ max_peak = attack_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 ((buf_peak[i] == 0) && (max_peak == 0)) continue;
+ /* if new peak, start attack slope */
+ if (buf_peak[i] >= max_peak)
+ {
+ max_peak = buf_peak[i];
+ }
+ /* keep sloping */
+ else
+ {
+ if (max_peak > attack_slope)
+ max_peak -= attack_slope;
+ else
+ max_peak = 0;
+ 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 ((buf_peak[i] == 0) && (release_peak == 0)) continue;
+ /* if new peak, start release slope */
+ if (buf_peak[i] >= release_peak)
+ {
+ release_peak = buf_peak[i];
+ }
+ /* keep sloping */
+ else
+ {
+ release_peak--;
+ 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 < 2; ch++)
+ {
+ int32_t *d = buf[ch];
+ for (i = 0; i < count; i++)
+ {
+ gain_peak = (buf_peak[i] + 1) / 90;
+ gain_rem = (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);
+ }
+ }
+}
Index: apps/settings_list.c
===================================================================
--- apps/settings_list.c (revision 21906)
+++ apps/settings_list.c (working copy)
@@ -1236,6 +1236,12 @@
/* 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,
Index: apps/dsp.h
===================================================================
--- apps/dsp.h (revision 21906)
+++ apps/dsp.h (working copy)
@@ -26,6 +26,8 @@
#include
#define NATIVE_FREQUENCY 44100
+#define ATTACK_BUFFER_SIZE 288 /* ~6.5 ms */
+#define MAX_LIMITER_GAIN 80 /* 8 dB */
enum
{
STEREO_INTERLEAVED = 0,
@@ -53,7 +55,11 @@
DSP_SET_ALBUM_GAIN,
DSP_SET_TRACK_PEAK,
DSP_SET_ALBUM_PEAK,
- DSP_CROSSFEED
+ DSP_CROSSFEED,
+ DSP_RESET_NO_ATTACK_BUFFER /* Disables the attack buffer. Reset the DSP
+ with this if calling dsp_process() without
+ calling both dsp_output_count() and
+ dsp_input_count() first */
};
enum {
@@ -88,5 +94,7 @@
void dsp_set_timestretch(int32_t percent);
int32_t dsp_get_timestretch(void);
int dsp_callback(int msg, intptr_t param);
+int dsp_flush_attack_buffer(char *dest);
+void dsp_set_limiter(int limiter_level);
#endif