forked from len0rd/rockbox
SWCODEC & Coldfire: Do some more DSP straigntening out. Do as much Coldfire optimizing as seems reasonably possible by jumping through some hoops to avoid stalls. Further boost reduction will just be fractional points if taken to extremes-- not worth it. Wrap up the ASM for awhile.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12905 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Michael Sevakis
parent
cd630c9e0a
commit
369c2a37b7
3 changed files with 469 additions and 370 deletions
354
apps/dsp.c
354
apps/dsp.c
|
|
@ -38,9 +38,14 @@
|
|||
#define WORD_FRACBITS 27
|
||||
|
||||
#define NATIVE_DEPTH 16
|
||||
/* If the buffer sizes change, check the assembly code! */
|
||||
#define SAMPLE_BUF_COUNT 256
|
||||
#define RESAMPLE_BUF_COUNT (256 * 4) /* Enough for 11,025 Hz -> 44,100 Hz*/
|
||||
#define DEFAULT_GAIN 0x01000000
|
||||
#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 (RESAMPLE_BUF_COUNT/2)
|
||||
|
||||
/* enums to index conversion properly with stereo mode and other settings */
|
||||
enum
|
||||
|
|
@ -66,11 +71,10 @@ enum
|
|||
* NOTE: Any assembly routines that use these structures must be updated
|
||||
* if current data members are moved or changed.
|
||||
*/
|
||||
/* 32-bit achitecture offset */
|
||||
struct resample_data
|
||||
{
|
||||
long delta; /* 00h */
|
||||
long phase; /* 04h */
|
||||
uint32_t delta; /* 00h */
|
||||
uint32_t phase; /* 04h */
|
||||
int32_t last_sample[2]; /* 08h */
|
||||
/* 10h */
|
||||
};
|
||||
|
|
@ -93,9 +97,10 @@ struct dsp_data
|
|||
int output_scale; /* 00h */
|
||||
int num_channels; /* 04h */
|
||||
struct resample_data resample_data; /* 08h */
|
||||
int clip_min; /* 18h */
|
||||
int clip_max; /* 2ch */
|
||||
/* 30h */
|
||||
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 */
|
||||
|
|
@ -132,13 +137,18 @@ struct eq_state
|
|||
#include <dsp_asm.h>
|
||||
|
||||
/* Typedefs keep things much neater in this case */
|
||||
typedef int (*sample_input_fn_type)(int count, const char *src[],
|
||||
int32_t *dst[]);
|
||||
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,
|
||||
int32_t *src[], int32_t *dst[]);
|
||||
typedef void (*sample_output_fn_type)(int count, struct dsp_data *data,
|
||||
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[]);
|
||||
|
||||
|
||||
/*
|
||||
***************************************************************************/
|
||||
|
|
@ -152,16 +162,16 @@ struct dsp_config
|
|||
int sample_bytes;
|
||||
int stereo_mode;
|
||||
int frac_bits;
|
||||
long gain; /* Note that this is in S8.23 format. */
|
||||
/* 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;
|
||||
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_fn_type apply_crossfeed;
|
||||
channels_process_fn_type channels_process;
|
||||
channels_process_dsp_fn_type apply_gain;
|
||||
channels_process_fn_type apply_crossfeed;
|
||||
channels_process_fn_type channels_process;
|
||||
};
|
||||
|
||||
/* General DSP config */
|
||||
|
|
@ -211,7 +221,7 @@ static struct dsp_config *dsp IDATA_ATTR = audio_dsp;
|
|||
* of copying needed is minimized for that case.
|
||||
*/
|
||||
|
||||
static int32_t sample_buf[SAMPLE_BUF_COUNT] IBSS_ATTR;
|
||||
int32_t sample_buf[SAMPLE_BUF_COUNT] IBSS_ATTR;
|
||||
static int32_t resample_buf[RESAMPLE_BUF_COUNT] IBSS_ATTR;
|
||||
|
||||
/* set a new dsp and return old one */
|
||||
|
|
@ -258,23 +268,20 @@ void sound_set_pitch(int permille)
|
|||
dsp_configure(DSP_SWITCH_FREQUENCY, dsp->codec_frequency);
|
||||
}
|
||||
|
||||
/* Convert at most count samples to the internal format, if needed. Returns
|
||||
* number of samples ready for further processing. 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 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 int sample_input_lte_native_mono(
|
||||
static void sample_input_lte_native_mono(
|
||||
int count, const char *src[], int32_t *dst[])
|
||||
{
|
||||
count = MIN(SAMPLE_BUF_COUNT/2, count);
|
||||
|
||||
const int16_t *s = (int16_t *) src[0];
|
||||
const int16_t * const send = s + count;
|
||||
int32_t *d = dst[0] = dst[1] = sample_buf;
|
||||
const int scale = WORD_SHIFT;
|
||||
int32_t *d = dst[0] = dst[1] = &sample_buf[SAMPLE_BUF_LEFT_CHANNEL];
|
||||
int scale = WORD_SHIFT;
|
||||
|
||||
do
|
||||
{
|
||||
|
|
@ -283,21 +290,17 @@ static int sample_input_lte_native_mono(
|
|||
while (s < send);
|
||||
|
||||
src[0] = (char *)s;
|
||||
|
||||
return count;
|
||||
}
|
||||
|
||||
/* convert count 16-bit interleaved stereo to 32-bit noninterleaved */
|
||||
static int sample_input_lte_native_i_stereo(
|
||||
static void sample_input_lte_native_i_stereo(
|
||||
int count, const char *src[], int32_t *dst[])
|
||||
{
|
||||
count = MIN(SAMPLE_BUF_COUNT/2, count);
|
||||
|
||||
const int32_t *s = (int32_t *) src[0];
|
||||
const int32_t * const send = s + count;
|
||||
int32_t *dl = dst[0] = sample_buf;
|
||||
int32_t *dr = dst[1] = sample_buf + SAMPLE_BUF_COUNT/2;
|
||||
const int scale = WORD_SHIFT;
|
||||
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;
|
||||
|
||||
do
|
||||
{
|
||||
|
|
@ -313,22 +316,18 @@ static int sample_input_lte_native_i_stereo(
|
|||
while (s < send);
|
||||
|
||||
src[0] = (char *)s;
|
||||
|
||||
return count;
|
||||
}
|
||||
|
||||
/* convert count 16-bit noninterleaved stereo to 32-bit noninterleaved */
|
||||
static int sample_input_lte_native_ni_stereo(
|
||||
static void sample_input_lte_native_ni_stereo(
|
||||
int count, const char *src[], int32_t *dst[])
|
||||
{
|
||||
count = MIN(SAMPLE_BUF_COUNT/2, count);
|
||||
|
||||
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;
|
||||
int32_t *dr = dst[1] = sample_buf + SAMPLE_BUF_COUNT/2;
|
||||
const int scale = WORD_SHIFT;
|
||||
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;
|
||||
|
||||
do
|
||||
{
|
||||
|
|
@ -339,35 +338,24 @@ static int sample_input_lte_native_ni_stereo(
|
|||
|
||||
src[0] = (char *)sl;
|
||||
src[1] = (char *)sr;
|
||||
|
||||
return count;
|
||||
}
|
||||
|
||||
/* convert count 32-bit mono to 32-bit mono */
|
||||
static int sample_input_gt_native_mono(
|
||||
static void sample_input_gt_native_mono(
|
||||
int count, const char *src[], int32_t *dst[])
|
||||
{
|
||||
count = MIN(SAMPLE_BUF_COUNT/2, count);
|
||||
|
||||
dst[0] = dst[1] = (int32_t *)src[0];
|
||||
src[0] = (char *)(dst[0] + count);
|
||||
|
||||
return count;
|
||||
}
|
||||
|
||||
/* convert count 32-bit interleaved stereo to 32-bit noninterleaved stereo */
|
||||
static int sample_input_gt_native_i_stereo(
|
||||
static void sample_input_gt_native_i_stereo(
|
||||
int count, const char *src[], int32_t *dst[])
|
||||
{
|
||||
count = MIN(SAMPLE_BUF_COUNT/2, count);
|
||||
|
||||
const int32_t *s = (int32_t *)src[0];
|
||||
const int32_t * const send = s + 2*count;
|
||||
int32_t *dl = sample_buf;
|
||||
int32_t *dr = sample_buf + SAMPLE_BUF_COUNT/2;
|
||||
|
||||
dst[0] = dl;
|
||||
dst[1] = dr;
|
||||
int32_t *dl = dst[0] = &sample_buf[SAMPLE_BUF_LEFT_CHANNEL];
|
||||
int32_t *dr = dst[1] = &sample_buf[SAMPLE_BUF_RIGHT_CHANNEL];
|
||||
|
||||
do
|
||||
{
|
||||
|
|
@ -377,22 +365,16 @@ static int sample_input_gt_native_i_stereo(
|
|||
while (s < send);
|
||||
|
||||
src[0] = (char *)send;
|
||||
|
||||
return count;
|
||||
}
|
||||
|
||||
/* convert 32 bit-noninterleaved stereo to 32-bit noninterleaved stereo */
|
||||
static int sample_input_gt_native_ni_stereo(
|
||||
static void sample_input_gt_native_ni_stereo(
|
||||
int count, const char *src[], int32_t *dst[])
|
||||
{
|
||||
count = MIN(SAMPLE_BUF_COUNT/2, count);
|
||||
|
||||
dst[0] = (int32_t *)src[0];
|
||||
dst[1] = (int32_t *)src[1];
|
||||
src[0] = (char *)(dst[0] + count);
|
||||
src[1] = (char *)(dst[1] + count);
|
||||
|
||||
return count;
|
||||
}
|
||||
|
||||
/**
|
||||
|
|
@ -573,12 +555,6 @@ static void sample_output_new_format(void)
|
|||
dsp->output_samples = sample_output_functions[out];
|
||||
}
|
||||
|
||||
static void resampler_set_delta(int frequency)
|
||||
{
|
||||
dsp->data.resample_data.delta = (unsigned long)
|
||||
frequency * 65536LL / NATIVE_FREQUENCY;
|
||||
}
|
||||
|
||||
/**
|
||||
* Linear interpolation resampling that introduces a one sample delay because
|
||||
* of our inability to look into the future at the end of a frame.
|
||||
|
|
@ -587,9 +563,9 @@ static void resampler_set_delta(int frequency)
|
|||
static int dsp_downsample(int count, struct dsp_data *data,
|
||||
int32_t *src[], int32_t *dst[])
|
||||
{
|
||||
int ch = data->num_channels - 1;
|
||||
long delta = data->resample_data.delta;
|
||||
long phase, pos;
|
||||
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. */
|
||||
|
|
@ -610,7 +586,7 @@ static int dsp_downsample(int count, struct dsp_data *data,
|
|||
if (pos > 0)
|
||||
last = s[pos - 1];
|
||||
|
||||
while (pos < count)
|
||||
while (pos < (uint32_t)count)
|
||||
{
|
||||
*d++ = last + FRACMUL((phase & 0xffff) << 15, s[pos] - last);
|
||||
phase += delta;
|
||||
|
|
@ -625,12 +601,12 @@ static int dsp_downsample(int count, struct dsp_data *data,
|
|||
return d - dst[0];
|
||||
}
|
||||
|
||||
static int dsp_upsample(int count, struct dsp_data *data,
|
||||
static int dsp_upsample(int count, struct dsp_data *data,
|
||||
int32_t *src[], int32_t *dst[])
|
||||
{
|
||||
int ch = data->num_channels - 1;
|
||||
long delta = data->resample_data.delta;
|
||||
long phase, pos;
|
||||
uint32_t delta = data->resample_data.delta;
|
||||
uint32_t phase, pos;
|
||||
int32_t *d;
|
||||
|
||||
/* Rolled channel loop actually showed slightly faster. */
|
||||
|
|
@ -653,7 +629,7 @@ static int dsp_upsample(int count, struct dsp_data *data,
|
|||
pos = phase >> 16;
|
||||
}
|
||||
|
||||
while (pos < count)
|
||||
while (pos < (uint32_t)count)
|
||||
{
|
||||
last = s[pos - 1];
|
||||
*d++ = last + FRACMUL((phase & 0xffff) << 15, s[pos] - last);
|
||||
|
|
@ -669,24 +645,43 @@ static int dsp_upsample(int count, struct dsp_data *data,
|
|||
}
|
||||
#endif /* DSP_HAVE_ASM_RESAMPLING */
|
||||
|
||||
static void resampler_new_delta(void)
|
||||
{
|
||||
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(int count, int32_t *src[])
|
||||
{
|
||||
if (dsp->resample)
|
||||
int32_t *dst[2] =
|
||||
{
|
||||
int32_t *dst[2] =
|
||||
{
|
||||
resample_buf,
|
||||
resample_buf + RESAMPLE_BUF_COUNT/2,
|
||||
};
|
||||
&resample_buf[RESAMPLE_BUF_LEFT_CHANNEL],
|
||||
&resample_buf[RESAMPLE_BUF_RIGHT_CHANNEL],
|
||||
};
|
||||
|
||||
count = dsp->resample(count, &dsp->data, src, dst);
|
||||
src[0] = dst[0];
|
||||
src[1] = dst[dsp->data.num_channels - 1];
|
||||
}
|
||||
count = dsp->resample(count, &dsp->data, src, dst);
|
||||
|
||||
src[0] = dst[0];
|
||||
src[1] = dst[dsp->data.num_channels - 1];
|
||||
|
||||
return count;
|
||||
}
|
||||
|
|
@ -810,30 +805,59 @@ void dsp_set_crossfeed_cross_params(long lf_gain, long hf_gain, long cutoff)
|
|||
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 = data->num_channels - 1;
|
||||
|
||||
do
|
||||
{
|
||||
int32_t *s = buf[ch];
|
||||
int32_t *d = buf[ch];
|
||||
int32_t samp = *s++;
|
||||
int i = 0;
|
||||
|
||||
do
|
||||
{
|
||||
FRACMUL_8_LOOP(samp, gain, s, d);
|
||||
}
|
||||
while (++i < count);
|
||||
}
|
||||
while (--ch >= 0);
|
||||
}
|
||||
#endif /* DSP_HAVE_ASM_APPLY_GAIN */
|
||||
|
||||
/* Combine all gains to a global gain. */
|
||||
static void set_gain(struct dsp_config *dsp)
|
||||
{
|
||||
dsp->gain = DEFAULT_GAIN;
|
||||
dsp->data.gain = DEFAULT_GAIN;
|
||||
|
||||
/* Replay gain not relevant to voice */
|
||||
if (dsp == audio_dsp && replaygain)
|
||||
{
|
||||
dsp->gain = replaygain;
|
||||
dsp->data.gain = replaygain;
|
||||
}
|
||||
|
||||
if (eq_enabled && eq_precut)
|
||||
{
|
||||
dsp->gain = (long) (((int64_t) dsp->gain * eq_precut) >> 24);
|
||||
dsp->data.gain =
|
||||
(long) (((int64_t) dsp->data.gain * eq_precut) >> 24);
|
||||
}
|
||||
|
||||
if (dsp->gain == DEFAULT_GAIN)
|
||||
if (dsp->data.gain == DEFAULT_GAIN)
|
||||
{
|
||||
dsp->gain = 0;
|
||||
dsp->data.gain = 0;
|
||||
}
|
||||
else
|
||||
{
|
||||
dsp->gain >>= 1;
|
||||
dsp->data.gain >>= 1;
|
||||
}
|
||||
|
||||
dsp->apply_gain = dsp->data.gain != 0 ? dsp_apply_gain : NULL;
|
||||
}
|
||||
|
||||
/**
|
||||
|
|
@ -927,50 +951,6 @@ static void eq_process(int count, int32_t *buf[])
|
|||
}
|
||||
}
|
||||
|
||||
/* Apply a constant gain to the samples (e.g., for ReplayGain). May update
|
||||
* the src array if gain was applied.
|
||||
* Note that this must be called before the resampler.
|
||||
*/
|
||||
static void apply_gain(int count, int32_t *buf[])
|
||||
{
|
||||
int32_t *sl, *sr;
|
||||
int32_t s, *d;
|
||||
long gain;
|
||||
int i;
|
||||
|
||||
if (new_gain)
|
||||
{
|
||||
/* Gain has changed */
|
||||
dsp_set_replaygain();
|
||||
if (dsp->gain == 0)
|
||||
return; /* No gain to apply now */
|
||||
}
|
||||
|
||||
sl = buf[0], sr = buf[1];
|
||||
gain = dsp->gain;
|
||||
|
||||
if (sl != sr)
|
||||
{
|
||||
d = &sample_buf[SAMPLE_BUF_COUNT / 2];
|
||||
buf[1] = d;
|
||||
s = *sr++;
|
||||
|
||||
for (i = 0; i < count; i++)
|
||||
FRACMUL_8_LOOP(s, gain, sr, d);
|
||||
}
|
||||
else
|
||||
{
|
||||
buf[1] = &sample_buf[0];
|
||||
}
|
||||
|
||||
d = &sample_buf[0];
|
||||
buf[0] = d;
|
||||
s = *sl++;
|
||||
|
||||
for (i = 0; i < count; i++)
|
||||
FRACMUL_8_LOOP(s, gain, sl, d);
|
||||
}
|
||||
|
||||
void dsp_set_stereo_width(int value)
|
||||
{
|
||||
long width, straight, cross;
|
||||
|
|
@ -993,35 +973,6 @@ void dsp_set_stereo_width(int value)
|
|||
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];
|
||||
|
||||
do
|
||||
{
|
||||
int32_t lr = *sl/2 + *sr/2;
|
||||
*sl++ = lr;
|
||||
*sr++ = lr;
|
||||
}
|
||||
while (--count > 0);
|
||||
}
|
||||
#endif /* DSP_HAVE_ASM_SOUND_CHAN_MONO */
|
||||
|
||||
#if CONFIG_CODEC == SWCODEC
|
||||
|
||||
#ifdef HAVE_SW_TONE_CONTROLS
|
||||
|
|
@ -1063,6 +1014,35 @@ int dsp_callback(int msg, intptr_t param)
|
|||
}
|
||||
#endif
|
||||
|
||||
/**
|
||||
* 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];
|
||||
|
||||
do
|
||||
{
|
||||
int32_t lr = *sl/2 + *sr/2;
|
||||
*sl++ = lr;
|
||||
*sr++ = lr;
|
||||
}
|
||||
while (--count > 0);
|
||||
}
|
||||
#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[])
|
||||
{
|
||||
|
|
@ -1151,30 +1131,47 @@ int dsp_process(char *dst, const char *src[], int count)
|
|||
coldfire_set_macsr(EMAC_FRACTIONAL | EMAC_SATURATE);
|
||||
#endif
|
||||
|
||||
if (new_gain)
|
||||
dsp_set_replaygain(); /* Gain has changed */
|
||||
|
||||
/* Testing function pointers for NULL is preferred since the pointer
|
||||
will be preloaded to be used for the call if not. */
|
||||
while (count > 0)
|
||||
{
|
||||
samples = dsp->input_samples(count, src, tmp);
|
||||
samples = MIN(SAMPLE_BUF_COUNT/2, count);
|
||||
count -= samples;
|
||||
if (dsp->gain != 0)
|
||||
apply_gain(samples, tmp);
|
||||
if ((samples = resample(samples, tmp)) <= 0)
|
||||
|
||||
dsp->input_samples(samples, src, tmp);
|
||||
|
||||
if (dsp->apply_gain)
|
||||
dsp->apply_gain(samples, &dsp->data, tmp);
|
||||
|
||||
if (dsp->resample && (samples = resample(samples, tmp)) <= 0)
|
||||
break; /* I'm pretty sure we're downsampling here */
|
||||
|
||||
if (dsp->apply_crossfeed)
|
||||
dsp->apply_crossfeed(samples, tmp);
|
||||
|
||||
/* TODO: EQ and tone controls need separate structs for audio and voice
|
||||
* DSP processing thanks to filter history. isn't really audible now, but
|
||||
* might be the day we start handling voice more delicately.
|
||||
* might be the day we start handling voice more delicately. Planned
|
||||
* changes may well run all relevent channels through the same EQ so
|
||||
* perhaps not.
|
||||
*/
|
||||
if (eq_enabled)
|
||||
eq_process(samples, tmp);
|
||||
|
||||
#ifdef HAVE_SW_TONE_CONTROLS
|
||||
if ((bass | treble) != 0)
|
||||
eq_filter(tmp, &tone_filter, samples, dsp->data.num_channels,
|
||||
FILTER_BISHELF_SHIFT);
|
||||
#endif
|
||||
|
||||
if (dsp->channels_process)
|
||||
dsp->channels_process(samples, tmp);
|
||||
|
||||
dsp->output_samples(samples, &dsp->data, tmp, (int16_t *)dst);
|
||||
|
||||
written += samples;
|
||||
dst += samples * sizeof (int16_t) * 2;
|
||||
yield();
|
||||
|
|
@ -1245,9 +1242,6 @@ bool dsp_configure(int setting, intptr_t value)
|
|||
if (dsp == audio_dsp)
|
||||
{
|
||||
*var = value;
|
||||
/* In case current gain is zero, force at least one call
|
||||
to apply_gain or apply_gain won't pick up on new_gain */
|
||||
audio_dsp->gain = -1;
|
||||
new_gain = true;
|
||||
}
|
||||
}
|
||||
|
|
@ -1282,15 +1276,7 @@ bool dsp_configure(int setting, intptr_t value)
|
|||
else
|
||||
dsp->frequency = dsp->codec_frequency;
|
||||
|
||||
resampler_set_delta(dsp->frequency);
|
||||
|
||||
if (dsp->frequency == NATIVE_FREQUENCY)
|
||||
dsp->resample = NULL;
|
||||
else if (dsp->frequency < NATIVE_FREQUENCY)
|
||||
dsp->resample = dsp_upsample;
|
||||
else
|
||||
dsp->resample = dsp_downsample;
|
||||
|
||||
resampler_new_delta();
|
||||
break;
|
||||
|
||||
case DSP_SET_SAMPLE_DEPTH:
|
||||
|
|
@ -1348,7 +1334,7 @@ bool dsp_configure(int setting, intptr_t value)
|
|||
case DSP_FLUSH:
|
||||
memset(&dsp->data.resample_data, 0,
|
||||
sizeof (dsp->data.resample_data));
|
||||
resampler_set_delta(dsp->frequency);
|
||||
resampler_new_delta();
|
||||
dither_init();
|
||||
break;
|
||||
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue