devs/foxbox
1
0
Fork 0
forked from len0rd/rockbox
Code Pull requests Activity

Remove pointless IRAM allocation from voice DSP.

Browse source 
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
This commit is contained in:
Michael Sevakis 2012-05-04 22:00:44 -04:00
parent dbe5e5f2df
commit 88aeef9127
4 changed files with 67 additions and 14 deletions
Download patch file Download diff file Expand all files Collapse all files

3
firmware/export/pcm_mixer.h
View file

@ -40,8 +40,7 @@
#define MIX_FRAME_SAMPLES 256
#endif
/* IAUDIO_M5 is very tight on IRAM */
#if (defined(CPU_COLDFIRE) && !defined(IAUDIO_M5)) || defined(CPU_PP)
#if defined(CPU_COLDFIRE) || defined(CPU_PP)
/* For Coldfire, it's just faster
For PortalPlayer, this also avoids more expensive cache coherency */
#define DOWNMIX_BUF_IBSS IBSS_ATTR

31
lib/rbcodec/dsp/dsp_sample_input.c
View file

@ -51,6 +51,10 @@
extern void dsp_sample_output_init(struct sample_io_data *this);
extern void dsp_sample_output_flush(struct sample_io_data *this);
#define SAMPLE_BUF_COUNT 128 /* Per channel, per DSP */
/* CODEC_IDX_AUDIO = left and right, CODEC_IDX_VOICE = mono */
static int32_t sample_bufs[3][SAMPLE_BUF_COUNT] IBSS_ATTR;
/* convert count 16-bit mono to 32-bit mono */
static void sample_input_mono16(struct sample_io_data *this,
struct dsp_buffer **buf_p)
@ -269,8 +273,31 @@ static void dsp_sample_input_format_change(struct sample_io_data *this,
format_change_ack(&src->format);
}
static void dsp_sample_input_init(struct sample_io_data *this)
static void dsp_sample_input_init(struct sample_io_data *this,
enum dsp_ids dsp_id)
{
int32_t *lbuf, *rbuf;
switch (dsp_id)
{
case CODEC_IDX_AUDIO:
lbuf = sample_bufs[0];
rbuf = sample_bufs[1];
break;
case CODEC_IDX_VOICE:
lbuf = rbuf = sample_bufs[2]; /* Always mono */
break;
default:
/* orly */
DEBUGF("DSP Input- unknown dsp %d\n", (int)dsp_id);
return;
}
this->sample_buf_arr[0] = lbuf;
this->sample_buf_arr[1] = rbuf;
this->input_samples[0] = sample_input_ni_stereo32;
this->input_samples[1] = dsp_sample_input_format_change;
}
@ -288,7 +315,7 @@ void dsp_sample_io_configure(struct sample_io_data *this,
switch (setting)
{
case DSP_INIT:
dsp_sample_input_init(this);
dsp_sample_input_init(this, (enum dsp_ids)value);
dsp_sample_output_init(this);
break;

4
lib/rbcodec/dsp/dsp_sample_io.h
View file

@ -28,8 +28,6 @@
#define WORD_FRACBITS 27
#define NATIVE_DEPTH 16
#define SAMPLE_BUF_COUNT 128 /* Per channel, per DSP */
struct sample_io_data;
/* DSP initial buffer input function call prototype */
@ -50,7 +48,7 @@ struct sample_io_data
int stereo_mode; /* Codec-specified input format */
sample_input_fn_type input_samples[2]; /* input functions */
struct dsp_buffer sample_buf; /* Buffer descriptor for converted samples */
int32_t sample_buf_arr[2][SAMPLE_BUF_COUNT]; /* Internal format */
int32_t *sample_buf_arr[2]; /* Internal format buffer pointers */
sample_output_fn_type output_samples[2]; /* Final output functions */
};

43
lib/rbcodec/dsp/lin_resample.c
View file

@ -40,6 +40,9 @@
#define RESAMPLE_BUF_COUNT 192 /* Per channel, per DSP */
/* CODEC_IDX_AUDIO = left and right, CODEC_IDX_VOICE = mono */
static int32_t resample_out_bufs[3][RESAMPLE_BUF_COUNT] IBSS_ATTR;
/* Data for each resampler on each DSP */
static struct resample_data
{
@ -50,7 +53,7 @@ static struct resample_data
/* 14h */
struct dsp_config *dsp; /* The DSP for this resampler */
struct dsp_buffer resample_buf; /* Buffer descriptor for resampled data */
int32_t resample_buf_arr[2][RESAMPLE_BUF_COUNT]; /* Actual output data */
int32_t *resample_buf_arr[2]; /* Actual output data pointers */
} resample_data[DSP_COUNT] IBSS_ATTR;
/* Actual worker function. Implemented here or in target assembly code. */
@ -165,11 +168,9 @@ static void lin_resample_process(struct dsp_proc_entry *this,
if (dst->remcount > 0)
return; /* data still remains */
int channels = src->format.num_channels;
dst->remcount = 0;
dst->p32[0] = data->resample_buf_arr[0];
dst->p32[1] = data->resample_buf_arr[channels - 1];
dst->p32[1] = data->resample_buf_arr[1];
if (src->remcount > 0)
{
@ -238,6 +239,36 @@ static void lin_resample_new_format(struct dsp_proc_entry *this,
dsp_proc_call(this, buf_p, 0);
}
static void lin_resample_init(struct dsp_config *dsp,
enum dsp_ids dsp_id)
{
/* Always enable resampler so that format changes may be monitored and
* it self-activated when required */
dsp_proc_enable(dsp, DSP_PROC_RESAMPLE, true);
int32_t *lbuf, *rbuf;
switch (dsp_id)
{
case CODEC_IDX_AUDIO:
lbuf = resample_out_bufs[0];
rbuf = resample_out_bufs[1];
break;
case CODEC_IDX_VOICE:
lbuf = rbuf = resample_out_bufs[2]; /* Always mono */
break;
default:
/* huh? */
DEBUGF("DSP_PROC_RESAMPLE- unknown DSP %d\n", (int)dsp_id);
return;
}
resample_data[dsp_id].resample_buf_arr[0] = lbuf;
resample_data[dsp_id].resample_buf_arr[1] = rbuf;
}
/* DSP message hook */
static intptr_t lin_resample_configure(struct dsp_proc_entry *this,
struct dsp_config *dsp,
@ -247,9 +278,7 @@ static intptr_t lin_resample_configure(struct dsp_proc_entry *this,
switch (setting)
{
case DSP_INIT:
/* Always enable resampler so that format changes may be monitored and
* it self-activated when required */
dsp_proc_enable(dsp, DSP_PROC_RESAMPLE, true);
lin_resample_init(dsp, (enum dsp_ids)value);
break;
case DSP_FLUSH: