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rockbox/apps/dsp.c
Jeffrey Goode 582225967f Tweak logf statements 
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@23486 a1c6a512-1295-4272-9138-f99709370657
2009-11-02 15:50:56 +00:00

1852 lines
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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Miika Pekkarinen
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include "config.h"
#include <stdbool.h>
#include <inttypes.h>
#include <string.h>
#include <sound.h>
#include "dsp.h"
#include "eq.h"
#include "kernel.h"
#include "system.h"
#include "settings.h"
#include "replaygain.h"
#include "tdspeed.h"
#include "buffer.h"
#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.
*/
#define WORD_SHIFT 12
#define WORD_FRACBITS 27
#define NATIVE_DEPTH 16
/* If the small buffer size changes, check the assembly code! */
#define SMALL_SAMPLE_BUF_COUNT 256
#define DEFAULT_GAIN 0x01000000
/* enums to index conversion properly with stereo mode and other settings */
enum
{
SAMPLE_INPUT_LE_NATIVE_I_STEREO = STEREO_INTERLEAVED,
SAMPLE_INPUT_LE_NATIVE_NI_STEREO = STEREO_NONINTERLEAVED,
SAMPLE_INPUT_LE_NATIVE_MONO = STEREO_MONO,
SAMPLE_INPUT_GT_NATIVE_I_STEREO = STEREO_INTERLEAVED + STEREO_NUM_MODES,
SAMPLE_INPUT_GT_NATIVE_NI_STEREO = STEREO_NONINTERLEAVED + STEREO_NUM_MODES,
SAMPLE_INPUT_GT_NATIVE_MONO = STEREO_MONO + STEREO_NUM_MODES,
SAMPLE_INPUT_GT_NATIVE_1ST_INDEX = STEREO_NUM_MODES
};
enum
{
SAMPLE_OUTPUT_MONO = 0,
SAMPLE_OUTPUT_STEREO,
SAMPLE_OUTPUT_DITHERED_MONO,
SAMPLE_OUTPUT_DITHERED_STEREO
};
/****************************************************************************
* NOTE: Any assembly routines that use these structures must be updated
* if current data members are moved or changed.
*/
struct resample_data
{
uint32_t delta; /* 00h */
uint32_t phase; /* 04h */
int32_t last_sample[2]; /* 08h */
/* 10h */
};
/* This is for passing needed data to assembly dsp routines. If another
* dsp parameter needs to be passed, add to the end of the structure
* and remove from dsp_config.
* If another function type becomes assembly optimized and requires dsp
* config info, add a pointer paramter of type "struct dsp_data *".
* If removing something from other than the end, reserve the spot or
* else update every implementation for every target.
* Be sure to add the offset of the new member for easy viewing as well. :)
* It is the first member of dsp_config and all members can be accessesed
* through the main aggregate but this is intended to make a safe haven
* for these items whereas the c part can be rearranged at will. dsp_data
* could even moved within dsp_config without disurbing the order.
*/
struct dsp_data
{
int output_scale; /* 00h */
int num_channels; /* 04h */
struct resample_data resample_data; /* 08h */
int32_t clip_min; /* 18h */
int32_t clip_max; /* 1ch */
int32_t gain; /* 20h - Note that this is in S8.23 format. */
/* 24h */
};
/* No asm...yet */
struct dither_data
{
long error[3]; /* 00h */
long random; /* 0ch */
/* 10h */
};
struct crossfeed_data
{
int32_t gain; /* 00h - Direct path gain */
int32_t coefs[3]; /* 04h - Coefficients for the shelving filter */
int32_t history[4]; /* 10h - Format is x[n - 1], y[n - 1] for both channels */
int32_t delay[13][2]; /* 20h */
int32_t *index; /* 88h - Current pointer into the delay line */
/* 8ch */
};
/* Current setup is one lowshelf filters three peaking filters and one
* highshelf filter. Varying the number of shelving filters make no sense,
* but adding peaking filters is possible.
*/
struct eq_state
{
char enabled[5]; /* 00h - Flags for active filters */
struct eqfilter filters[5]; /* 08h - packing is 4? */
/* 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>
/* Typedefs keep things much neater in this case */
typedef void (*sample_input_fn_type)(int count, const char *src[],
int32_t *dst[]);
typedef int (*resample_fn_type)(int count, struct dsp_data *data,
const int32_t *src[], int32_t *dst[]);
typedef void (*sample_output_fn_type)(int count, struct dsp_data *data,
const int32_t *src[], int16_t *dst);
/* Single-DSP channel processing in place */
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[]);
/*
***************************************************************************/
struct dsp_config
{
struct dsp_data data; /* Config members for use in asm routines */
long codec_frequency; /* Sample rate of data coming from the codec */
long frequency; /* Effective sample rate after pitch shift (if any) */
int sample_depth;
int sample_bytes;
int stereo_mode;
int32_t tdspeed_percent; /* Speed% * PITCH_SPEED_PRECISION */
bool tdspeed_active; /* Timestretch is in use */
int frac_bits;
#ifdef HAVE_SW_TONE_CONTROLS
/* Filter struct for software bass/treble controls */
struct eqfilter tone_filter;
#endif
/* Functions that change depending upon settings - NULL if stage is
disabled */
sample_input_fn_type input_samples;
resample_fn_type resample;
sample_output_fn_type output_samples;
/* These will be NULL for the voice codec and is more economical that
way */
channels_process_dsp_fn_type apply_gain;
channels_process_fn_type apply_crossfeed;
channels_process_fn_type eq_process;
channels_process_fn_type channels_process;
channels_process_fn_type compressor_process;
};
/* General DSP config */
static struct dsp_config dsp_conf[2] IBSS_ATTR; /* 0=A, 1=V */
/* Dithering */
static struct dither_data dither_data[2] IBSS_ATTR; /* 0=left, 1=right */
static long dither_mask IBSS_ATTR;
static long dither_bias IBSS_ATTR;
/* Crossfeed */
struct crossfeed_data crossfeed_data IDATA_ATTR = /* A */
{
.index = (int32_t *)crossfeed_data.delay
};
/* Equalizer */
static struct eq_state eq_data; /* A */
/* Software tone controls */
#ifdef HAVE_SW_TONE_CONTROLS
static int prescale; /* A/V */
static int bass; /* A/V */
static int treble; /* A/V */
#endif
/* Settings applicable to audio codec only */
static int32_t pitch_ratio = PITCH_SPEED_100;
static int channels_mode;
long dsp_sw_gain;
long dsp_sw_cross;
static bool dither_enabled;
static long eq_precut;
static long track_gain;
static bool new_gain;
static long album_gain;
static long track_peak;
static long album_peak;
static long replaygain;
static bool crossfeed_enabled;
#define AUDIO_DSP (dsp_conf[CODEC_IDX_AUDIO])
#define VOICE_DSP (dsp_conf[CODEC_IDX_VOICE])
/* The internal format is 32-bit samples, non-interleaved, stereo. This
* format is similar to the raw output from several codecs, so the amount
* of copying needed is minimized for that case.
*/
#define RESAMPLE_RATIO 4 /* Enough for 11,025 Hz -> 44,100 Hz */
static int32_t small_sample_buf[SMALL_SAMPLE_BUF_COUNT] IBSS_ATTR;
static int32_t small_resample_buf[SMALL_SAMPLE_BUF_COUNT * RESAMPLE_RATIO] IBSS_ATTR;
static int32_t *big_sample_buf = NULL;
static int32_t *big_resample_buf = NULL;
static int big_sample_buf_count = -1; /* -1=unknown, 0=not available */
static int sample_buf_count;
static int32_t *sample_buf;
static int32_t *resample_buf;
#define SAMPLE_BUF_LEFT_CHANNEL 0
#define SAMPLE_BUF_RIGHT_CHANNEL (sample_buf_count/2)
#define RESAMPLE_BUF_LEFT_CHANNEL 0
#define RESAMPLE_BUF_RIGHT_CHANNEL (sample_buf_count/2 * RESAMPLE_RATIO)
/* compressor */
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[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 */
static inline int32_t clip_sample_16(int32_t sample)
{
if ((int16_t)sample != sample)
sample = 0x7fff ^ (sample >> 31);
return sample;
}
int32_t sound_get_pitch(void)
{
return pitch_ratio;
}
void sound_set_pitch(int32_t percent)
{
pitch_ratio = percent;
dsp_configure(&AUDIO_DSP, DSP_SWITCH_FREQUENCY,
AUDIO_DSP.codec_frequency);
}
static void tdspeed_setup(struct dsp_config *dspc)
{
/* Assume timestretch will not be used */
dspc->tdspeed_active = false;
sample_buf = small_sample_buf;
resample_buf = small_resample_buf;
sample_buf_count = SMALL_SAMPLE_BUF_COUNT;
if(!dsp_timestretch_available())
return; /* Timestretch not enabled or buffer not allocated */
if (dspc->tdspeed_percent == 0)
dspc->tdspeed_percent = PITCH_SPEED_100;
if (!tdspeed_config(
dspc->codec_frequency == 0 ? NATIVE_FREQUENCY : dspc->codec_frequency,
dspc->stereo_mode != STEREO_MONO,
dspc->tdspeed_percent))
return; /* Timestretch not possible or needed with these parameters */
/* Timestretch is to be used */
dspc->tdspeed_active = true;
sample_buf = big_sample_buf;
sample_buf_count = big_sample_buf_count;
resample_buf = big_resample_buf;
}
void dsp_timestretch_enable(bool enabled)
{
/* Hook to set up timestretch buffer on first call to settings_apply() */
if (big_sample_buf_count < 0) /* Only do something on first call */
{
if (enabled)
{
/* Set up timestretch buffers */
big_sample_buf_count = SMALL_SAMPLE_BUF_COUNT * RESAMPLE_RATIO;
big_sample_buf = small_resample_buf;
big_resample_buf = (int32_t *) buffer_alloc(big_sample_buf_count * RESAMPLE_RATIO * sizeof(int32_t));
}
else
{
/* Not enabled at startup, "big" buffers will never be available */
big_sample_buf_count = 0;
}
tdspeed_setup(&AUDIO_DSP);
}
}
void dsp_set_timestretch(int32_t percent)
{
AUDIO_DSP.tdspeed_percent = percent;
tdspeed_setup(&AUDIO_DSP);
}
int32_t dsp_get_timestretch()
{
return AUDIO_DSP.tdspeed_percent;
}
bool dsp_timestretch_available()
{
return (global_settings.timestretch_enabled && big_sample_buf_count > 0);
}
/* Convert count samples to the internal format, if needed. Updates src
* to point past the samples "consumed" and dst is set to point to the
* samples to consume. Note that for mono, dst[0] equals dst[1], as there
* is no point in processing the same data twice.
*/
/* convert count 16-bit mono to 32-bit mono */
static void sample_input_lte_native_mono(
int count, const char *src[], int32_t *dst[])
{
const int16_t *s = (int16_t *) src[0];
const int16_t * const send = s + count;
int32_t *d = dst[0] = dst[1] = &sample_buf[SAMPLE_BUF_LEFT_CHANNEL];
int scale = WORD_SHIFT;
while (s < send)
{
*d++ = *s++ << scale;
}
src[0] = (char *)s;
}
/* convert count 16-bit interleaved stereo to 32-bit noninterleaved */
static void sample_input_lte_native_i_stereo(
int count, const char *src[], int32_t *dst[])
{
const int32_t *s = (int32_t *) src[0];
const int32_t * const send = s + count;
int32_t *dl = dst[0] = &sample_buf[SAMPLE_BUF_LEFT_CHANNEL];
int32_t *dr = dst[1] = &sample_buf[SAMPLE_BUF_RIGHT_CHANNEL];
int scale = WORD_SHIFT;
while (s < send)
{
int32_t slr = *s++;
#ifdef ROCKBOX_LITTLE_ENDIAN
*dl++ = (slr >> 16) << scale;
*dr++ = (int32_t)(int16_t)slr << scale;
#else /* ROCKBOX_BIG_ENDIAN */
*dl++ = (int32_t)(int16_t)slr << scale;
*dr++ = (slr >> 16) << scale;
#endif
}
src[0] = (char *)s;
}
/* convert count 16-bit noninterleaved stereo to 32-bit noninterleaved */
static void sample_input_lte_native_ni_stereo(
int count, const char *src[], int32_t *dst[])
{
const int16_t *sl = (int16_t *) src[0];
const int16_t *sr = (int16_t *) src[1];
const int16_t * const slend = sl + count;
int32_t *dl = dst[0] = &sample_buf[SAMPLE_BUF_LEFT_CHANNEL];
int32_t *dr = dst[1] = &sample_buf[SAMPLE_BUF_RIGHT_CHANNEL];
int scale = WORD_SHIFT;
while (sl < slend)
{
*dl++ = *sl++ << scale;
*dr++ = *sr++ << scale;
}
src[0] = (char *)sl;
src[1] = (char *)sr;
}
/* convert count 32-bit mono to 32-bit mono */
static void sample_input_gt_native_mono(
int count, const char *src[], int32_t *dst[])
{
dst[0] = dst[1] = (int32_t *)src[0];
src[0] = (char *)(dst[0] + count);
}
/* convert count 32-bit interleaved stereo to 32-bit noninterleaved stereo */
static void sample_input_gt_native_i_stereo(
int count, const char *src[], int32_t *dst[])
{
const int32_t *s = (int32_t *)src[0];
const int32_t * const send = s + 2*count;
int32_t *dl = dst[0] = &sample_buf[SAMPLE_BUF_LEFT_CHANNEL];
int32_t *dr = dst[1] = &sample_buf[SAMPLE_BUF_RIGHT_CHANNEL];
while (s < send)
{
*dl++ = *s++;
*dr++ = *s++;
}
src[0] = (char *)send;
}
/* convert 32 bit-noninterleaved stereo to 32-bit noninterleaved stereo */
static void sample_input_gt_native_ni_stereo(
int count, const char *src[], int32_t *dst[])
{
dst[0] = (int32_t *)src[0];
dst[1] = (int32_t *)src[1];
src[0] = (char *)(dst[0] + count);
src[1] = (char *)(dst[1] + count);
}
/**
* sample_input_new_format()
*
* set the to-native sample conversion function based on dsp sample parameters
*
* !DSPPARAMSYNC
* needs syncing with changes to the following dsp parameters:
* * dsp->stereo_mode (A/V)
* * dsp->sample_depth (A/V)
*/
static void sample_input_new_format(struct dsp_config *dsp)
{
static const sample_input_fn_type sample_input_functions[] =
{
[SAMPLE_INPUT_LE_NATIVE_I_STEREO] = sample_input_lte_native_i_stereo,
[SAMPLE_INPUT_LE_NATIVE_NI_STEREO] = sample_input_lte_native_ni_stereo,
[SAMPLE_INPUT_LE_NATIVE_MONO] = sample_input_lte_native_mono,
[SAMPLE_INPUT_GT_NATIVE_I_STEREO] = sample_input_gt_native_i_stereo,
[SAMPLE_INPUT_GT_NATIVE_NI_STEREO] = sample_input_gt_native_ni_stereo,
[SAMPLE_INPUT_GT_NATIVE_MONO] = sample_input_gt_native_mono,
};
int convert = dsp->stereo_mode;
if (dsp->sample_depth > NATIVE_DEPTH)
convert += SAMPLE_INPUT_GT_NATIVE_1ST_INDEX;
dsp->input_samples = sample_input_functions[convert];
}
#ifndef DSP_HAVE_ASM_SAMPLE_OUTPUT_MONO
/* write mono internal format to output format */
static void sample_output_mono(int count, struct dsp_data *data,
const int32_t *src[], int16_t *dst)
{
const int32_t *s0 = src[0];
const int scale = data->output_scale;
const int dc_bias = 1 << (scale - 1);
while (count-- > 0)
{
int32_t lr = clip_sample_16((*s0++ + dc_bias) >> scale);
*dst++ = lr;
*dst++ = lr;
}
}
#endif /* DSP_HAVE_ASM_SAMPLE_OUTPUT_MONO */
/* write stereo internal format to output format */
#ifndef DSP_HAVE_ASM_SAMPLE_OUTPUT_STEREO
static void sample_output_stereo(int count, struct dsp_data *data,
const int32_t *src[], int16_t *dst)
{
const int32_t *s0 = src[0];
const int32_t *s1 = src[1];
const int scale = data->output_scale;
const int dc_bias = 1 << (scale - 1);
while (count-- > 0)
{
*dst++ = clip_sample_16((*s0++ + dc_bias) >> scale);
*dst++ = clip_sample_16((*s1++ + dc_bias) >> scale);
}
}
#endif /* DSP_HAVE_ASM_SAMPLE_OUTPUT_STEREO */
/**
* The "dither" code to convert the 24-bit samples produced by libmad was
* taken from the coolplayer project - coolplayer.sourceforge.net
*
* This function handles mono and stereo outputs.
*/
static void sample_output_dithered(int count, struct dsp_data *data,
const int32_t *src[], int16_t *dst)
{
const int32_t mask = dither_mask;
const int32_t bias = dither_bias;
const int scale = data->output_scale;
const int32_t min = data->clip_min;
const int32_t max = data->clip_max;
const int32_t range = max - min;
int ch;
int16_t *d;
for (ch = 0; ch < data->num_channels; ch++)
{
struct dither_data * const dither = &dither_data[ch];
const int32_t *s = src[ch];
int i;
for (i = 0, d = &dst[ch]; i < count; i++, s++, d += 2)
{
int32_t output, sample;
int32_t random;
/* Noise shape and bias (for correct rounding later) */
sample = *s;
sample += dither->error[0] - dither->error[1] + dither->error[2];
dither->error[2] = dither->error[1];
dither->error[1] = dither->error[0]/2;
output = sample + bias;
/* Dither, highpass triangle PDF */
random = dither->random*0x0019660dL + 0x3c6ef35fL;
output += (random & mask) - (dither->random & mask);
dither->random = random;
/* Round sample to output range */
output &= ~mask;
/* Error feedback */
dither->error[0] = sample - output;
/* Clip */
if ((uint32_t)(output - min) > (uint32_t)range)
{
int32_t c = min;
if (output > min)
c += range;
output = c;
}
/* Quantize and store */
*d = output >> scale;
}
}
if (data->num_channels == 2)
return;
/* Have to duplicate left samples into the right channel since
pcm buffer and hardware is interleaved stereo */
d = &dst[0];
while (count-- > 0)
{
int16_t s = *d++;
*d++ = s;
}
}
/**
* sample_output_new_format()
*
* set the from-native to ouput sample conversion routine
*
* !DSPPARAMSYNC
* needs syncing with changes to the following dsp parameters:
* * dsp->stereo_mode (A/V)
* * dither_enabled (A)
*/
static void sample_output_new_format(struct dsp_config *dsp)
{
static const sample_output_fn_type sample_output_functions[] =
{
sample_output_mono,
sample_output_stereo,
sample_output_dithered,
sample_output_dithered
};
int out = dsp->data.num_channels - 1;
if (dsp == &AUDIO_DSP && dither_enabled)
out += 2;
dsp->output_samples = sample_output_functions[out];
}
/**
* Linear interpolation resampling that introduces a one sample delay because
* of our inability to look into the future at the end of a frame.
*/
#ifndef DSP_HAVE_ASM_RESAMPLING
static int dsp_downsample(int count, struct dsp_data *data,
const int32_t *src[], int32_t *dst[])
{
int ch = data->num_channels - 1;
uint32_t delta = data->resample_data.delta;
uint32_t phase, pos;
int32_t *d;
/* Rolled channel loop actually showed slightly faster. */
do
{
/* Just initialize things and not worry too much about the relatively
* uncommon case of not being able to spit out a sample for the frame.
*/
const int32_t *s = src[ch];
int32_t last = data->resample_data.last_sample[ch];
data->resample_data.last_sample[ch] = s[count - 1];
d = dst[ch];
phase = data->resample_data.phase;
pos = phase >> 16;
/* Do we need last sample of previous frame for interpolation? */
if (pos > 0)
last = s[pos - 1];
while (pos < (uint32_t)count)
{
*d++ = last + FRACMUL((phase & 0xffff) << 15, s[pos] - last);
phase += delta;
pos = phase >> 16;
last = s[pos - 1];
}
}
while (--ch >= 0);
/* Wrap phase accumulator back to start of next frame. */
data->resample_data.phase = phase - (count << 16);
return d - dst[0];
}
static int dsp_upsample(int count, struct dsp_data *data,
const int32_t *src[], int32_t *dst[])
{
int ch = data->num_channels - 1;
uint32_t delta = data->resample_data.delta;
uint32_t phase, pos;
int32_t *d;
/* Rolled channel loop actually showed slightly faster. */
do
{
/* Should always be able to output a sample for a ratio up to RESAMPLE_RATIO */
const int32_t *s = src[ch];
int32_t last = data->resample_data.last_sample[ch];
data->resample_data.last_sample[ch] = s[count - 1];
d = dst[ch];
phase = data->resample_data.phase;
pos = phase >> 16;
while (pos == 0)
{
*d++ = last + FRACMUL((phase & 0xffff) << 15, s[0] - last);
phase += delta;
pos = phase >> 16;
}
while (pos < (uint32_t)count)
{
last = s[pos - 1];
*d++ = last + FRACMUL((phase & 0xffff) << 15, s[pos] - last);
phase += delta;
pos = phase >> 16;
}
}
while (--ch >= 0);
/* Wrap phase accumulator back to start of next frame. */
data->resample_data.phase = phase & 0xffff;
return d - dst[0];
}
#endif /* DSP_HAVE_ASM_RESAMPLING */
static void resampler_new_delta(struct dsp_config *dsp)
{
dsp->data.resample_data.delta = (unsigned long)
dsp->frequency * 65536LL / NATIVE_FREQUENCY;
if (dsp->frequency == NATIVE_FREQUENCY)
{
/* NOTE: If fully glitch-free transistions from no resampling to
resampling are desired, last_sample history should be maintained
even when not resampling. */
dsp->resample = NULL;
dsp->data.resample_data.phase = 0;
dsp->data.resample_data.last_sample[0] = 0;
dsp->data.resample_data.last_sample[1] = 0;
}
else if (dsp->frequency < NATIVE_FREQUENCY)
dsp->resample = dsp_upsample;
else
dsp->resample = dsp_downsample;
}
/* Resample count stereo samples. Updates the src array, if resampling is
* done, to refer to the resampled data. Returns number of stereo samples
* for further processing.
*/
static inline int resample(struct dsp_config *dsp, int count, int32_t *src[])
{
int32_t *dst[2] =
{
&resample_buf[RESAMPLE_BUF_LEFT_CHANNEL],
&resample_buf[RESAMPLE_BUF_RIGHT_CHANNEL],
};
count = dsp->resample(count, &dsp->data, (const int32_t **)src, dst);
src[0] = dst[0];
src[1] = dst[dsp->data.num_channels - 1];
return count;
}
static void dither_init(struct dsp_config *dsp)
{
memset(dither_data, 0, sizeof (dither_data));
dither_bias = (1L << (dsp->frac_bits - NATIVE_DEPTH));
dither_mask = (1L << (dsp->frac_bits + 1 - NATIVE_DEPTH)) - 1;
}
void dsp_dither_enable(bool enable)
{
struct dsp_config *dsp = &AUDIO_DSP;
dither_enabled = enable;
sample_output_new_format(dsp);
}
/* Applies crossfeed to the stereo signal in src.
* Crossfeed is a process where listening over speakers is simulated. This
* is good for old hard panned stereo records, which might be quite fatiguing
* to listen to on headphones with no crossfeed.
*/
#ifndef DSP_HAVE_ASM_CROSSFEED
static void apply_crossfeed(int count, int32_t *buf[])
{
int32_t *hist_l = &crossfeed_data.history[0];
int32_t *hist_r = &crossfeed_data.history[2];
int32_t *delay = &crossfeed_data.delay[0][0];
int32_t *coefs = &crossfeed_data.coefs[0];
int32_t gain = crossfeed_data.gain;
int32_t *di = crossfeed_data.index;
int32_t acc;
int32_t left, right;
int i;
for (i = 0; i < count; i++)
{
left = buf[0][i];
right = buf[1][i];
/* Filter delayed sample from left speaker */
acc = FRACMUL(*di, coefs[0]);
acc += FRACMUL(hist_l[0], coefs[1]);
acc += FRACMUL(hist_l[1], coefs[2]);
/* Save filter history for left speaker */
hist_l[1] = acc;
hist_l[0] = *di;
*di++ = left;
/* Filter delayed sample from right speaker */
acc = FRACMUL(*di, coefs[0]);
acc += FRACMUL(hist_r[0], coefs[1]);
acc += FRACMUL(hist_r[1], coefs[2]);
/* Save filter history for right speaker */
hist_r[1] = acc;
hist_r[0] = *di;
*di++ = right;
/* Now add the attenuated direct sound and write to outputs */
buf[0][i] = FRACMUL(left, gain) + hist_r[1];
buf[1][i] = FRACMUL(right, gain) + hist_l[1];
/* Wrap delay line index if bigger than delay line size */
if (di >= delay + 13*2)
di = delay;
}
/* Write back local copies of data we've modified */
crossfeed_data.index = di;
}
#endif /* DSP_HAVE_ASM_CROSSFEED */
/**
* dsp_set_crossfeed(bool enable)
*
* !DSPPARAMSYNC
* needs syncing with changes to the following dsp parameters:
* * dsp->stereo_mode (A)
*/
void dsp_set_crossfeed(bool enable)
{
crossfeed_enabled = enable;
AUDIO_DSP.apply_crossfeed = (enable && AUDIO_DSP.data.num_channels > 1)
? apply_crossfeed : NULL;
}
void dsp_set_crossfeed_direct_gain(int gain)
{
crossfeed_data.gain = get_replaygain_int(gain * 10) << 7;
/* If gain is negative, the calculation overflowed and we need to clamp */
if (crossfeed_data.gain < 0)
crossfeed_data.gain = 0x7fffffff;
}
/* Both gains should be below 0 dB */
void dsp_set_crossfeed_cross_params(long lf_gain, long hf_gain, long cutoff)
{
int32_t *c = crossfeed_data.coefs;
long scaler = get_replaygain_int(lf_gain * 10) << 7;
cutoff = 0xffffffff/NATIVE_FREQUENCY*cutoff;
hf_gain -= lf_gain;
/* Divide cutoff by sqrt(10^(hf_gain/20)) to place cutoff at the -3 dB
* point instead of shelf midpoint. This is for compatibility with the old
* crossfeed shelf filter and should be removed if crossfeed settings are
* ever made incompatible for any other good reason.
*/
cutoff = fp_div(cutoff, get_replaygain_int(hf_gain*5), 24);
filter_shelf_coefs(cutoff, hf_gain, false, c);
/* Scale coefs by LF gain and shift them to s0.31 format. We have no gains
* over 1 and can do this safely
*/
c[0] = FRACMUL_SHL(c[0], scaler, 4);
c[1] = FRACMUL_SHL(c[1], scaler, 4);
c[2] <<= 4;
}
/* Apply a constant gain to the samples (e.g., for ReplayGain).
* Note that this must be called before the resampler.
*/
#ifndef DSP_HAVE_ASM_APPLY_GAIN
static void dsp_apply_gain(int count, struct dsp_data *data, int32_t *buf[])
{
const int32_t gain = data->gain;
int ch;
for (ch = 0; ch < data->num_channels; ch++)
{
int32_t *d = buf[ch];
int i;
for (i = 0; i < count; i++)
d[i] = FRACMUL_SHL(d[i], gain, 8);
}
}
#endif /* DSP_HAVE_ASM_APPLY_GAIN */
/* 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 */
if (dsp == &AUDIO_DSP && replaygain)
{
dsp->data.gain = replaygain;
}
if (dsp->eq_process && eq_precut)
{
dsp->data.gain = fp_mul(dsp->data.gain, eq_precut, 24);
}
#ifdef HAVE_SW_VOLUME_CONTROL
if (global_settings.volume < SW_VOLUME_MAX ||
global_settings.volume > SW_VOLUME_MIN)
{
int vol_gain = get_replaygain_int(global_settings.volume * 100);
dsp->data.gain = (long) (((int64_t) dsp->data.gain * vol_gain) >> 24);
}
#endif
if (dsp->data.gain == DEFAULT_GAIN)
{
dsp->data.gain = 0;
}
else
{
dsp->data.gain >>= 1; /* convert gain to S8.23 format */
}
dsp->apply_gain = dsp->data.gain != 0 ? dsp_apply_gain : NULL;
}
/**
* Update the amount to cut the audio before applying the equalizer.
*
* @param precut to apply in decibels (multiplied by 10)
*/
void dsp_set_eq_precut(int precut)
{
eq_precut = get_replaygain_int(precut * -10);
set_gain(&AUDIO_DSP);
}
/**
* Synchronize the equalizer filter coefficients with the global settings.
*
* @param band the equalizer band to synchronize
*/
void dsp_set_eq_coefs(int band)
{
const int *setting;
long gain;
unsigned long cutoff, q;
/* Adjust setting pointer to the band we actually want to change */
setting = &global_settings.eq_band0_cutoff + (band * 3);
/* Convert user settings to format required by coef generator functions */
cutoff = 0xffffffff / NATIVE_FREQUENCY * (*setting++);
q = *setting++;
gain = *setting++;
if (q == 0)
q = 1;
/* NOTE: The coef functions assume the EMAC unit is in fractional mode,
which it should be, since we're executed from the main thread. */
/* Assume a band is disabled if the gain is zero */
if (gain == 0)
{
eq_data.enabled[band] = 0;
}
else
{
if (band == 0)
eq_ls_coefs(cutoff, q, gain, eq_data.filters[band].coefs);
else if (band == 4)
eq_hs_coefs(cutoff, q, gain, eq_data.filters[band].coefs);
else
eq_pk_coefs(cutoff, q, gain, eq_data.filters[band].coefs);
eq_data.enabled[band] = 1;
}
}
/* Apply EQ filters to those bands that have got it switched on. */
static void eq_process(int count, int32_t *buf[])
{
static const int shifts[] =
{
EQ_SHELF_SHIFT, /* low shelf */
EQ_PEAK_SHIFT, /* peaking */
EQ_PEAK_SHIFT, /* peaking */
EQ_PEAK_SHIFT, /* peaking */
EQ_SHELF_SHIFT, /* high shelf */
};
unsigned int channels = AUDIO_DSP.data.num_channels;
int i;
/* filter configuration currently is 1 low shelf filter, 3 band peaking
filters and 1 high shelf filter, in that order. we need to know this
so we can choose the correct shift factor.
*/
for (i = 0; i < 5; i++)
{
if (!eq_data.enabled[i])
continue;
eq_filter(buf, &eq_data.filters[i], count, channels, shifts[i]);
}
}
/**
* Use to enable the equalizer.
*
* @param enable true to enable the equalizer
*/
void dsp_set_eq(bool enable)
{
AUDIO_DSP.eq_process = enable ? eq_process : NULL;
set_gain(&AUDIO_DSP);
}
static void dsp_set_stereo_width(int value)
{
long width, straight, cross;
width = value * 0x7fffff / 100;
if (value <= 100)
{
straight = (0x7fffff + width) / 2;
cross = straight - width;
}
else
{
/* straight = (1 + width) / (2 * width) */
straight = ((int64_t)(0x7fffff + width) << 22) / width;
cross = straight - 0x7fffff;
}
dsp_sw_gain = straight << 8;
dsp_sw_cross = cross << 8;
}
/**
* Implements the different channel configurations and stereo width.
*/
/* SOUND_CHAN_STEREO mode is a noop so has no function - just outline one for
* completeness. */
#if 0
static void channels_process_sound_chan_stereo(int count, int32_t *buf[])
{
/* The channels are each just themselves */
(void)count; (void)buf;
}
#endif
#ifndef DSP_HAVE_ASM_SOUND_CHAN_MONO
static void channels_process_sound_chan_mono(int count, int32_t *buf[])
{
int32_t *sl = buf[0], *sr = buf[1];
while (count-- > 0)
{
int32_t lr = *sl/2 + *sr/2;
*sl++ = lr;
*sr++ = lr;
}
}
#endif /* DSP_HAVE_ASM_SOUND_CHAN_MONO */
#ifndef DSP_HAVE_ASM_SOUND_CHAN_CUSTOM
static void channels_process_sound_chan_custom(int count, int32_t *buf[])
{
const int32_t gain = dsp_sw_gain;
const int32_t cross = dsp_sw_cross;
int32_t *sl = buf[0], *sr = buf[1];
while (count-- > 0)
{
int32_t l = *sl;
int32_t r = *sr;
*sl++ = FRACMUL(l, gain) + FRACMUL(r, cross);
*sr++ = FRACMUL(r, gain) + FRACMUL(l, cross);
}
}
#endif /* DSP_HAVE_ASM_SOUND_CHAN_CUSTOM */
static void channels_process_sound_chan_mono_left(int count, int32_t *buf[])
{
/* Just copy over the other channel */
memcpy(buf[1], buf[0], count * sizeof (*buf));
}
static void channels_process_sound_chan_mono_right(int count, int32_t *buf[])
{
/* Just copy over the other channel */
memcpy(buf[0], buf[1], count * sizeof (*buf));
}
#ifndef DSP_HAVE_ASM_SOUND_CHAN_KARAOKE
static void channels_process_sound_chan_karaoke(int count, int32_t *buf[])
{
int32_t *sl = buf[0], *sr = buf[1];
while (count-- > 0)
{
int32_t ch = *sl/2 - *sr/2;
*sl++ = ch;
*sr++ = -ch;
}
}
#endif /* DSP_HAVE_ASM_SOUND_CHAN_KARAOKE */
static void dsp_set_channel_config(int value)
{
static const channels_process_fn_type channels_process_functions[] =
{
/* SOUND_CHAN_STEREO = All-purpose index for no channel processing */
[SOUND_CHAN_STEREO] = NULL,
[SOUND_CHAN_MONO] = channels_process_sound_chan_mono,
[SOUND_CHAN_CUSTOM] = channels_process_sound_chan_custom,
[SOUND_CHAN_MONO_LEFT] = channels_process_sound_chan_mono_left,
[SOUND_CHAN_MONO_RIGHT] = channels_process_sound_chan_mono_right,
[SOUND_CHAN_KARAOKE] = channels_process_sound_chan_karaoke,
};
if ((unsigned)value >= ARRAYLEN(channels_process_functions) ||
AUDIO_DSP.stereo_mode == STEREO_MONO)
{
value = SOUND_CHAN_STEREO;
}
/* This doesn't apply to voice */
channels_mode = value;
AUDIO_DSP.channels_process = channels_process_functions[value];
}
#if CONFIG_CODEC == SWCODEC
#ifdef HAVE_SW_TONE_CONTROLS
static void set_tone_controls(void)
{
filter_bishelf_coefs(0xffffffff/NATIVE_FREQUENCY*200,
0xffffffff/NATIVE_FREQUENCY*3500,
bass, treble, -prescale,
AUDIO_DSP.tone_filter.coefs);
/* Sync the voice dsp coefficients */
memcpy(&VOICE_DSP.tone_filter.coefs, AUDIO_DSP.tone_filter.coefs,
sizeof (VOICE_DSP.tone_filter.coefs));
}
#endif
/* Hook back from firmware/ part of audio, which can't/shouldn't call apps/
* code directly.
*/
int dsp_callback(int msg, intptr_t param)
{
switch (msg)
{
#ifdef HAVE_SW_TONE_CONTROLS
case DSP_CALLBACK_SET_PRESCALE:
prescale = param;
set_tone_controls();
break;
/* prescaler is always set after calling any of these, so we wait with
* calculating coefs until the above case is hit.
*/
case DSP_CALLBACK_SET_BASS:
bass = param;
break;
case DSP_CALLBACK_SET_TREBLE:
treble = param;
break;
#ifdef HAVE_SW_VOLUME_CONTROL
case DSP_CALLBACK_SET_SW_VOLUME:
set_gain(&AUDIO_DSP);
break;
#endif
#endif
case DSP_CALLBACK_SET_CHANNEL_CONFIG:
dsp_set_channel_config(param);
break;
case DSP_CALLBACK_SET_STEREO_WIDTH:
dsp_set_stereo_width(param);
break;
default:
break;
}
return 0;
}
#endif
/* Process and convert src audio to dst based on the DSP configuration,
* reading count number of audio samples. dst is assumed to be large
* enough; use dsp_output_count() to get the required number. src is an
* array of pointers; for mono and interleaved stereo, it contains one
* pointer to the start of the audio data and the other is ignored; for
* non-interleaved stereo, it contains two pointers, one for each audio
* channel. Returns number of bytes written to dst.
*/
int dsp_process(struct dsp_config *dsp, char *dst, const char *src[], int count)
{
int32_t *tmp[2];
static long last_yield;
long tick;
int written = 0;
#if defined(CPU_COLDFIRE)
/* set emac unit for dsp processing, and save old macsr, we're running in
codec thread context at this point, so can't clobber it */
unsigned long old_macsr = coldfire_get_macsr();
coldfire_set_macsr(EMAC_FRACTIONAL | EMAC_SATURATE);
#endif
if (new_gain)
dsp_set_replaygain(); /* Gain has changed */
/* Perform at least one yield before starting */
last_yield = current_tick;
yield();
/* Testing function pointers for NULL is preferred since the pointer
will be preloaded to be used for the call if not. */
while (count > 0)
{
int samples = MIN(sample_buf_count/2, count);
count -= samples;
dsp->input_samples(samples, src, tmp);
if (dsp->tdspeed_active)
samples = tdspeed_doit(tmp, samples);
int chunk_offset = 0;
while (samples > 0)
{
int32_t *t2[2];
t2[0] = tmp[0]+chunk_offset;
t2[1] = tmp[1]+chunk_offset;
int chunk = MIN(sample_buf_count/2, samples);
chunk_offset += chunk;
samples -= chunk;
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->apply_crossfeed)
dsp->apply_crossfeed(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);
#endif
if (dsp->channels_process)
dsp->channels_process(chunk, t2);
if (dsp->compressor_process)
dsp->compressor_process(chunk, t2);
dsp->output_samples(chunk, &dsp->data, (const int32_t **)t2, (int16_t *)dst);
written += chunk;
dst += chunk * sizeof (int16_t) * 2;
/* yield at least once each tick */
tick = current_tick;
if (TIME_AFTER(tick, last_yield))
{
last_yield = tick;
yield();
}
}
}
#if defined(CPU_COLDFIRE)
/* set old macsr again */
coldfire_set_macsr(old_macsr);
#endif
return written;
}
/* Given count number of input samples, calculate the maximum number of
* samples of output data that would be generated (the calculation is not
* entirely exact and rounds upwards to be on the safe side; during
* resampling, the number of samples generated depends on the current state
* of the resampler).
*/
/* dsp_input_size MUST be called afterwards */
int dsp_output_count(struct dsp_config *dsp, int count)
{
if (dsp->tdspeed_active)
count = tdspeed_est_output_size();
if (dsp->resample)
{
count = (int)(((unsigned long)count * NATIVE_FREQUENCY
+ (dsp->frequency - 1)) / dsp->frequency);
}
/* Now we have the resampled sample count which must not exceed
* RESAMPLE_BUF_RIGHT_CHANNEL to avoid resample buffer overflow. One
* must call dsp_input_count() to get the correct input sample
* count.
*/
if (count > RESAMPLE_BUF_RIGHT_CHANNEL)
count = RESAMPLE_BUF_RIGHT_CHANNEL;
return count;
}
/* Given count output samples, calculate number of input samples
* that would be consumed in order to fill the output buffer.
*/
int dsp_input_count(struct dsp_config *dsp, int count)
{
/* count is now the number of resampled input samples. Convert to
original input samples. */
if (dsp->resample)
{
/* Use the real resampling delta =
* dsp->frequency * 65536 / NATIVE_FREQUENCY, and
* round towards zero to avoid buffer overflows. */
count = (int)(((unsigned long)count *
dsp->data.resample_data.delta) >> 16);
}
if (dsp->tdspeed_active)
count = tdspeed_est_input_size(count);
return count;
}
static void dsp_set_gain_var(long *var, long value)
{
*var = value;
new_gain = true;
}
static void dsp_update_functions(struct dsp_config *dsp)
{
sample_input_new_format(dsp);
sample_output_new_format(dsp);
if (dsp == &AUDIO_DSP)
dsp_set_crossfeed(crossfeed_enabled);
}
intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value)
{
switch (setting)
{
case DSP_MYDSP:
switch (value)
{
case CODEC_IDX_AUDIO:
return (intptr_t)&AUDIO_DSP;
case CODEC_IDX_VOICE:
return (intptr_t)&VOICE_DSP;
default:
return (intptr_t)NULL;
}
case DSP_SET_FREQUENCY:
memset(&dsp->data.resample_data, 0, sizeof (dsp->data.resample_data));
/* Fall through!!! */
case DSP_SWITCH_FREQUENCY:
dsp->codec_frequency = (value == 0) ? NATIVE_FREQUENCY : value;
/* Account for playback speed adjustment when setting dsp->frequency
if we're called from the main audio thread. Voice UI thread should
not need this feature.
*/
if (dsp == &AUDIO_DSP)
dsp->frequency = pitch_ratio * dsp->codec_frequency / PITCH_SPEED_100;
else
dsp->frequency = dsp->codec_frequency;
resampler_new_delta(dsp);
tdspeed_setup(dsp);
break;
case DSP_SET_SAMPLE_DEPTH:
dsp->sample_depth = value;
if (dsp->sample_depth <= NATIVE_DEPTH)
{
dsp->frac_bits = WORD_FRACBITS;
dsp->sample_bytes = sizeof (int16_t); /* samples are 16 bits */
dsp->data.clip_max = ((1 << WORD_FRACBITS) - 1);
dsp->data.clip_min = -((1 << WORD_FRACBITS));
}
else
{
dsp->frac_bits = value;
dsp->sample_bytes = sizeof (int32_t); /* samples are 32 bits */
dsp->data.clip_max = (1 << value) - 1;
dsp->data.clip_min = -(1 << value);
}
dsp->data.output_scale = dsp->frac_bits + 1 - NATIVE_DEPTH;
sample_input_new_format(dsp);
dither_init(dsp);
break;
case DSP_SET_STEREO_MODE:
dsp->stereo_mode = value;
dsp->data.num_channels = value == STEREO_MONO ? 1 : 2;
dsp_update_functions(dsp);
tdspeed_setup(dsp);
break;
case DSP_RESET:
dsp->stereo_mode = STEREO_NONINTERLEAVED;
dsp->data.num_channels = 2;
dsp->sample_depth = NATIVE_DEPTH;
dsp->frac_bits = WORD_FRACBITS;
dsp->sample_bytes = sizeof (int16_t);
dsp->data.output_scale = dsp->frac_bits + 1 - NATIVE_DEPTH;
dsp->data.clip_max = ((1 << WORD_FRACBITS) - 1);
dsp->data.clip_min = -((1 << WORD_FRACBITS));
dsp->codec_frequency = dsp->frequency = NATIVE_FREQUENCY;
if (dsp == &AUDIO_DSP)
{
track_gain = 0;
album_gain = 0;
track_peak = 0;
album_peak = 0;
new_gain = true;
}
dsp_update_functions(dsp);
resampler_new_delta(dsp);
tdspeed_setup(dsp);
if (dsp == &AUDIO_DSP)
release_gain = UNITY;
break;
case DSP_FLUSH:
memset(&dsp->data.resample_data, 0,
sizeof (dsp->data.resample_data));
resampler_new_delta(dsp);
dither_init(dsp);
tdspeed_setup(dsp);
if (dsp == &AUDIO_DSP)
release_gain = UNITY;
break;
case DSP_SET_TRACK_GAIN:
if (dsp == &AUDIO_DSP)
dsp_set_gain_var(&track_gain, value);
break;
case DSP_SET_ALBUM_GAIN:
if (dsp == &AUDIO_DSP)
dsp_set_gain_var(&album_gain, value);
break;
case DSP_SET_TRACK_PEAK:
if (dsp == &AUDIO_DSP)
dsp_set_gain_var(&track_peak, value);
break;
case DSP_SET_ALBUM_PEAK:
if (dsp == &AUDIO_DSP)
dsp_set_gain_var(&album_peak, value);
break;
default:
return 0;
}
return 1;
}
int get_replaygain_mode(bool have_track_gain, bool have_album_gain)
{
int type;
bool track = ((global_settings.replaygain_type == REPLAYGAIN_TRACK)
|| ((global_settings.replaygain_type == REPLAYGAIN_SHUFFLE)
&& global_settings.playlist_shuffle));
type = (!track && have_album_gain) ? REPLAYGAIN_ALBUM
: have_track_gain ? REPLAYGAIN_TRACK : -1;
return type;
}
void dsp_set_replaygain(void)
{
long gain = 0;
new_gain = false;
if ((global_settings.replaygain_type != REPLAYGAIN_OFF) ||
global_settings.replaygain_noclip)
{
bool track_mode = get_replaygain_mode(track_gain != 0,
album_gain != 0) == REPLAYGAIN_TRACK;
long peak = (track_mode || !album_peak) ? track_peak : album_peak;
if (global_settings.replaygain_type != REPLAYGAIN_OFF)
{
gain = (track_mode || !album_gain) ? track_gain : album_gain;
if (global_settings.replaygain_preamp)
{
long preamp = get_replaygain_int(
global_settings.replaygain_preamp * 10);
gain = (long) (((int64_t) gain * preamp) >> 24);
}
}
if (gain == 0)
{
/* So that noclip can work even with no gain information. */
gain = DEFAULT_GAIN;
}
if (global_settings.replaygain_noclip && (peak != 0)
&& ((((int64_t) gain * peak) >> 24) >= DEFAULT_GAIN))
{
gain = (((int64_t) DEFAULT_GAIN << 24) / peak);
}
if (gain == DEFAULT_GAIN)
{
/* Nothing to do, disable processing. */
gain = 0;
}
}
/* Store in S7.24 format to simplify calculations. */
replaygain = gain;
set_gain(&AUDIO_DSP);
}
/** SET COMPRESSOR
* Called by the menu system to configure the compressor process */
void dsp_set_compressor(int c_threshold, int c_ratio, int c_gain,
int c_knee, int c_release)
{
bool changed = false;
bool active = (c_threshold < 0);
const int comp_ratio[] = {2, 4, 6, 10, 0};
int new_ratio = comp_ratio[c_ratio];
bool new_knee = (c_knee == 1);
int new_release = c_release * NATIVE_FREQUENCY / 1000;
if (c_menu.threshold != c_threshold)
{
changed = true;
c_menu.threshold = c_threshold;
logf(" Compressor Threshold: %d dB\tEnabled: %s",
c_menu.threshold, active ? "Yes" : "No");
}
if (c_menu.ratio != new_ratio)
{
changed = true;
c_menu.ratio = new_ratio;
if (c_menu.ratio)
{
logf(" Compressor Ratio: %d:1", c_menu.ratio);
}
else
{
logf(" Compressor Ratio: Limit");
}
}
if (c_menu.gain != c_gain)
{
changed = true;
c_menu.gain = c_gain;
if (c_menu.gain >= 0)
{
logf(" Compressor Makeup Gain: %d dB", c_menu.gain);
}
else
{
logf(" Compressor Makeup Gain: Auto");
}
}
if (c_menu.soft_knee != new_knee)
{
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[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
[4] = ~+12db input (2 bits clipping overhead) */
db_curve[1].db = c_menu.threshold << 16;
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);
/* 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;
/* top of knee is at the threshold for hard knee */
db_curve[2].db = c_menu.threshold << 16;
db_curve[2].offset = 0;
}
/* 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] 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] = UNITY;
/* 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((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(ROCKBOX_HAS_LOGF) && defined(LOGF_ENABLE)
logf("\n *** Compression Offsets ***");
/* some settings for display only, not used in calculations */
db_curve[0].offset = 0;
db_curve[1].offset = 0;
db_curve[3].db = 0;
for (i = 0; i <= 4; i++)
{
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 <= 65; i++)
{
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 / 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 / UNITY);
release_gain = UNITY;
}
/* 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 inline int32_t get_compression_gain(int32_t sample)
{
const int frac_bits_offset = AUDIO_DSP.frac_bits - 15;
/* sample must be positive */
if (sample < 0)
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;
/* 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])));
}
/* sample is too clipped, return invalid value */
return -1;
}
/** COMPRESSOR PROCESS
* Changes the gain of the samples according to the compressor curve
*/
static void compressor_process(int count, int32_t *buf[])
{
const int num_chan = AUDIO_DSP.data.num_channels;
int32_t *in_buf[2] = {buf[0], buf[1]};
while (count-- > 0)
{
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++)
{
int32_t this_gain = get_compression_gain(*in_buf[ch]);
if (this_gain < sample_gain)
sample_gain = this_gain;
}
/* perform release slope; skip if no compression and no release slope */
if ((sample_gain != UNITY) || (release_gain != UNITY))
{
/* if larger offset than previous slope, start new release slope */
if ((sample_gain <= release_gain) && (sample_gain > 0))
{
release_gain = sample_gain;
}
else
/* keep sloping towards unity gain (and ignore invalid value) */
{
release_gain += comp_rel_slope;
if (release_gain > UNITY)
{
release_gain = UNITY;
}
}
}
/* 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++)
{
*in_buf[ch] = FRACMUL_SHL(total_gain, *in_buf[ch], 7);
}
}
in_buf[0]++;
in_buf[1]++;
}
}
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