Rockbox open source high quality audio player as a Music Player Daemon
mpris
rockbox
mpd
libadwaita
audio
rust
zig
deno
Remove pointless IRAM allocation from voice DSP.
It's always used in MONO mode and doesn't need the IRAM sample/
resample buffers and 1280 bytes can be freed.
M5 can now have its PCM mixer downmix buffer in IRAM.
Change-Id: I0af08be5b212b7dfe382bba588a6585eb328a038
+67
-14
+1
-2
firmware/export/pcm_mixer.h
+1
-2
firmware/export/pcm_mixer.h
···
40
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#define MIX_FRAME_SAMPLES 256
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#endif
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-
/* IAUDIO_M5 is very tight on IRAM */
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-
#if (defined(CPU_COLDFIRE) && !defined(IAUDIO_M5)) || defined(CPU_PP)
43
+
#if defined(CPU_COLDFIRE) || defined(CPU_PP)
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/* For Coldfire, it's just faster
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For PortalPlayer, this also avoids more expensive cache coherency */
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#define DOWNMIX_BUF_IBSS IBSS_ATTR
+29
-2
lib/rbcodec/dsp/dsp_sample_input.c
+29
-2
lib/rbcodec/dsp/dsp_sample_input.c
···
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extern void dsp_sample_output_init(struct sample_io_data *this);
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extern void dsp_sample_output_flush(struct sample_io_data *this);
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+
#define SAMPLE_BUF_COUNT 128 /* Per channel, per DSP */
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+
/* CODEC_IDX_AUDIO = left and right, CODEC_IDX_VOICE = mono */
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+
static int32_t sample_bufs[3][SAMPLE_BUF_COUNT] IBSS_ATTR;
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+
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/* convert count 16-bit mono to 32-bit mono */
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static void sample_input_mono16(struct sample_io_data *this,
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struct dsp_buffer **buf_p)
···
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format_change_ack(&src->format);
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}
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-
static void dsp_sample_input_init(struct sample_io_data *this)
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+
static void dsp_sample_input_init(struct sample_io_data *this,
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+
enum dsp_ids dsp_id)
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{
279
+
int32_t *lbuf, *rbuf;
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+
281
+
switch (dsp_id)
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+
{
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+
case CODEC_IDX_AUDIO:
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+
lbuf = sample_bufs[0];
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+
rbuf = sample_bufs[1];
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+
break;
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+
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+
case CODEC_IDX_VOICE:
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+
lbuf = rbuf = sample_bufs[2]; /* Always mono */
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+
break;
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+
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+
default:
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+
/* orly */
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DEBUGF("DSP Input- unknown dsp %d\n", (int)dsp_id);
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+
return;
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+
}
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+
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+
this->sample_buf_arr[0] = lbuf;
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+
this->sample_buf_arr[1] = rbuf;
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+
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this->input_samples[0] = sample_input_ni_stereo32;
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this->input_samples[1] = dsp_sample_input_format_change;
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}
···
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switch (setting)
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{
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case DSP_INIT:
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-
dsp_sample_input_init(this);
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+
dsp_sample_input_init(this, (enum dsp_ids)value);
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dsp_sample_output_init(this);
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break;
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+1
-3
lib/rbcodec/dsp/dsp_sample_io.h
+1
-3
lib/rbcodec/dsp/dsp_sample_io.h
···
28
28
#define WORD_FRACBITS 27
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29
#define NATIVE_DEPTH 16
30
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31
-
#define SAMPLE_BUF_COUNT 128 /* Per channel, per DSP */
32
-
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struct sample_io_data;
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33
/* DSP initial buffer input function call prototype */
···
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int stereo_mode; /* Codec-specified input format */
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sample_input_fn_type input_samples[2]; /* input functions */
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struct dsp_buffer sample_buf; /* Buffer descriptor for converted samples */
53
-
int32_t sample_buf_arr[2][SAMPLE_BUF_COUNT]; /* Internal format */
51
+
int32_t *sample_buf_arr[2]; /* Internal format buffer pointers */
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sample_output_fn_type output_samples[2]; /* Final output functions */
55
53
};
56
54
+36
-7
lib/rbcodec/dsp/lin_resample.c
+36
-7
lib/rbcodec/dsp/lin_resample.c
···
40
40
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#define RESAMPLE_BUF_COUNT 192 /* Per channel, per DSP */
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+
/* CODEC_IDX_AUDIO = left and right, CODEC_IDX_VOICE = mono */
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+
static int32_t resample_out_bufs[3][RESAMPLE_BUF_COUNT] IBSS_ATTR;
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+
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/* Data for each resampler on each DSP */
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static struct resample_data
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{
···
50
53
/* 14h */
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struct dsp_config *dsp; /* The DSP for this resampler */
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struct dsp_buffer resample_buf; /* Buffer descriptor for resampled data */
53
-
int32_t resample_buf_arr[2][RESAMPLE_BUF_COUNT]; /* Actual output data */
56
+
int32_t *resample_buf_arr[2]; /* Actual output data pointers */
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} resample_data[DSP_COUNT] IBSS_ATTR;
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/* Actual worker function. Implemented here or in target assembly code. */
···
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if (dst->remcount > 0)
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return; /* data still remains */
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168
-
int channels = src->format.num_channels;
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-
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dst->remcount = 0;
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dst->p32[0] = data->resample_buf_arr[0];
172
-
dst->p32[1] = data->resample_buf_arr[channels - 1];
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+
dst->p32[1] = data->resample_buf_arr[1];
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if (src->remcount > 0)
175
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{
···
238
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dsp_proc_call(this, buf_p, 0);
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}
240
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+
static void lin_resample_init(struct dsp_config *dsp,
243
+
enum dsp_ids dsp_id)
244
+
{
245
+
/* Always enable resampler so that format changes may be monitored and
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+
* it self-activated when required */
247
+
dsp_proc_enable(dsp, DSP_PROC_RESAMPLE, true);
248
+
249
+
int32_t *lbuf, *rbuf;
250
+
251
+
switch (dsp_id)
252
+
{
253
+
case CODEC_IDX_AUDIO:
254
+
lbuf = resample_out_bufs[0];
255
+
rbuf = resample_out_bufs[1];
256
+
break;
257
+
258
+
case CODEC_IDX_VOICE:
259
+
lbuf = rbuf = resample_out_bufs[2]; /* Always mono */
260
+
break;
261
+
262
+
default:
263
+
/* huh? */
264
+
DEBUGF("DSP_PROC_RESAMPLE- unknown DSP %d\n", (int)dsp_id);
265
+
return;
266
+
}
267
+
268
+
resample_data[dsp_id].resample_buf_arr[0] = lbuf;
269
+
resample_data[dsp_id].resample_buf_arr[1] = rbuf;
270
+
}
271
+
241
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/* DSP message hook */
242
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static intptr_t lin_resample_configure(struct dsp_proc_entry *this,
243
274
struct dsp_config *dsp,
···
247
278
switch (setting)
248
279
{
249
280
case DSP_INIT:
250
-
/* Always enable resampler so that format changes may be monitored and
251
-
* it self-activated when required */
252
-
dsp_proc_enable(dsp, DSP_PROC_RESAMPLE, true);
281
+
lin_resample_init(dsp, (enum dsp_ids)value);
253
282
break;
254
283
255
284
case DSP_FLUSH: