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rockbox/apps/dsp.c
Thomas Martitz baa070cca6 GSoC/Buflib: Enable compaction in buflib. 
This enables the ability to allocate (and free) memory dynamically
without fragmentation, through compaction. This means allocations can move
and fragmentation be reduced. Most changes are preparing Rockbox for this,
which many times means adding a move callback which can temporarily disable
movement when the corresponding code is in a critical section.

For now, the audio buffer allocation has a central role, because it's the one
having allocated most. This buffer is able to shrink itself, for which it
needs to stop playback for a very short moment. For this,
audio_buffer_available() returns the size of the audio buffer which can
possibly be used by other allocations because the audio buffer can shrink.

lastfm scrobbling and timestretch can now be toggled at runtime without
requiring a reboot.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30381 a1c6a512-1295-4272-9138-f99709370657
2011-08-30 14:01:45 +00:00

1892 lines
58 KiB
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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 "core_alloc.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 */
bool auto_gain; /* 0 = off, 1 = auto */
int ratio; /* 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 */
#ifdef HAVE_PITCHSCREEN
bool tdspeed_active; /* Timestretch is in use */
#endif
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 */
#ifdef HAVE_PITCHSCREEN
static int32_t pitch_ratio = PITCH_SPEED_100;
#endif
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;
#ifdef HAVE_PITCHSCREEN
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 */
#endif
static int sample_buf_count = SMALL_SAMPLE_BUF_COUNT;
static int32_t *sample_buf = small_sample_buf;
static int32_t *resample_buf = small_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;
}
#ifdef HAVE_PITCHSCREEN
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;
}
static int move_callback(int handle, void* current, void* new)
{
/* TODO */
(void)handle;(void)current;;
big_sample_buf = new;
return BUFLIB_CB_OK;
}
static struct buflib_callbacks ops = {
.move_callback = move_callback,
.shrink_callback = NULL,
};
void dsp_timestretch_enable(bool enabled)
{
/* Hook to set up timestretch buffer on first call to settings_apply() */
static int handle;
if (enabled)
{
if (big_sample_buf_count > 0)
return; /* already allocated and enabled */
/* Set up timestretch buffers */
big_sample_buf_count = SMALL_SAMPLE_BUF_COUNT * RESAMPLE_RATIO;
big_sample_buf = small_resample_buf;
handle = core_alloc_ex("resample buf",
big_sample_buf_count * RESAMPLE_RATIO * sizeof(int32_t), &ops);
if (handle > 0)
{ /* success, now setup tdspeed */
big_resample_buf = core_get_data(handle);
tdspeed_init();
tdspeed_setup(&AUDIO_DSP);
}
}
if (!enabled || (handle <= 0)) /* disable */
{
dsp_set_timestretch(PITCH_SPEED_100);
tdspeed_finish();
if (handle > 0)
core_free(handle);
handle = 0;
big_sample_buf = NULL;
big_sample_buf_count = 0;
}
}
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);
}
#endif
/* 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)
{
static int32_t *tmp[2]; /* tdspeed_doit() needs it static */
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);
#ifdef HAVE_PITCHSCREEN
if (dsp->tdspeed_active)
samples = tdspeed_doit(tmp, samples);
#endif
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)
{
#ifdef HAVE_PITCHSCREEN
if (dsp->tdspeed_active)
count = tdspeed_est_output_size();
#endif
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);
}
#ifdef HAVE_PITCHSCREEN
if (dsp->tdspeed_active)
count = tdspeed_est_input_size(count);
#endif
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.
*/
#ifdef HAVE_PITCHSCREEN
if (dsp == &AUDIO_DSP)
dsp->frequency = pitch_ratio * dsp->codec_frequency / PITCH_SPEED_100;
else
#endif
dsp->frequency = dsp->codec_frequency;
resampler_new_delta(dsp);
#ifdef HAVE_PITCHSCREEN
tdspeed_setup(dsp);
#endif
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);
#ifdef HAVE_PITCHSCREEN
tdspeed_setup(dsp);
#endif
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);
#ifdef HAVE_PITCHSCREEN
tdspeed_setup(dsp);
#endif
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);
#ifdef HAVE_PITCHSCREEN
tdspeed_setup(dsp);
#endif
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_gain, int c_ratio,
int c_knee, int c_release)
{
bool changed = false;
bool active = (c_threshold < 0);
bool new_auto_gain = (c_gain == 1);
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.auto_gain != new_auto_gain)
{
changed = true;
c_menu.auto_gain = new_auto_gain;
logf(" Compressor Makeup Gain: %s",
c_menu.auto_gain ? "Auto" : "Off");
}
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.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 */
comp_makeup_gain = c_menu.auto_gain ?
fp_factor(-(db_curve[3].offset) - 0x199A, 16) << 8 : UNITY;
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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