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

4 files changed, 67 insertions, 14 deletions

diff --git a/firmware/export/pcm_mixer.h b/firmware/export/pcm_mixer.h
index f6d212d8e0..69d2b894df 100644
--- a/firmware/export/pcm_mixer.h
+++ b/firmware/export/pcm_mixer.h
@@ -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
diff --git a/lib/rbcodec/dsp/dsp_sample_input.c b/lib/rbcodec/dsp/dsp_sample_input.c
index 97b4ec27ad..df0b01f8c6 100644
--- a/lib/rbcodec/dsp/dsp_sample_input.c
+++ b/lib/rbcodec/dsp/dsp_sample_input.c
@@ -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;
 
diff --git a/lib/rbcodec/dsp/dsp_sample_io.h b/lib/rbcodec/dsp/dsp_sample_io.h
index 443038919d..0afe75c6ca 100644
--- a/lib/rbcodec/dsp/dsp_sample_io.h
+++ b/lib/rbcodec/dsp/dsp_sample_io.h
@@ -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 */
 };
 
diff --git a/lib/rbcodec/dsp/lin_resample.c b/lib/rbcodec/dsp/lin_resample.c
index b3855ec768..5a6f55ac48 100644
--- a/lib/rbcodec/dsp/lin_resample.c
+++ b/lib/rbcodec/dsp/lin_resample.c
@@ -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: