Apply similar updates to audiofilters.Filter and audiodelays.Echo: MICROPY_FLOAT_CONST, MP_ROM_INT, and synthio_block_slot_get_limited.

relic-se · relic-se · commit 222ce2c78276 · 2024-12-10T09:33:35.000-06:00
diff --git a/shared-bindings/audiofilters/Filter.c b/shared-bindings/audiofilters/Filter.c
@@ -72,7 +72,7 @@ static mp_obj_t audiofilters_filter_make_new(const mp_obj_type_t *type, size_t n
     enum { ARG_filter, ARG_mix, ARG_buffer_size, ARG_sample_rate, ARG_bits_per_sample, ARG_samples_signed, ARG_channel_count, };
     static const mp_arg_t allowed_args[] = {
         { MP_QSTR_filter, MP_ARG_OBJ | MP_ARG_KW_ONLY, {.u_obj = MP_OBJ_NULL} },
-        { MP_QSTR_mix, MP_ARG_OBJ | MP_ARG_KW_ONLY,  {.u_obj = MP_OBJ_NULL} },
+        { MP_QSTR_mix, MP_ARG_OBJ | MP_ARG_KW_ONLY,  {.u_obj = MP_ROM_INT(1)} },
         { MP_QSTR_buffer_size, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 512} },
         { MP_QSTR_sample_rate, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 8000} },
         { MP_QSTR_bits_per_sample, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 16} },
diff --git a/shared-module/audiodelays/Echo.c b/shared-module/audiodelays/Echo.c
@@ -57,17 +57,17 @@ void common_hal_audiodelays_echo_construct(audiodelays_echo_obj_t *self, uint32_
 
     // If we did not receive a BlockInput we need to create a default float value
     if (decay == MP_OBJ_NULL) {
-        decay = mp_obj_new_float(0.7);
+        decay = mp_obj_new_float(MICROPY_FLOAT_CONST(0.7));
     }
     synthio_block_assign_slot(decay, &self->decay, MP_QSTR_decay);
 
     if (delay_ms == MP_OBJ_NULL) {
-        delay_ms = mp_obj_new_float(250.0);
+        delay_ms = mp_obj_new_float(MICROPY_FLOAT_CONST(250.0));
     }
     synthio_block_assign_slot(delay_ms, &self->delay_ms, MP_QSTR_delay_ms);
 
     if (mix == MP_OBJ_NULL) {
-        mix = mp_obj_new_float(0.5);
+        mix = mp_obj_new_float(MICROPY_FLOAT_CONST(0.5));
     }
     synthio_block_assign_slot(mix, &self->mix, MP_QSTR_mix);
 
@@ -77,7 +77,7 @@ void common_hal_audiodelays_echo_construct(audiodelays_echo_obj_t *self, uint32_
 
     // Allocate the echo buffer for the max possible delay, echo is always 16-bit
     self->max_delay_ms = max_delay_ms;
-    self->max_echo_buffer_len = (uint32_t)(self->sample_rate / 1000.0f * max_delay_ms) * (self->channel_count * sizeof(uint16_t)); // bytes
+    self->max_echo_buffer_len = (uint32_t)(self->sample_rate / MICROPY_FLOAT_CONST(1000.0) * max_delay_ms) * (self->channel_count * sizeof(uint16_t)); // bytes
     self->echo_buffer = m_malloc(self->max_echo_buffer_len);
     if (self->echo_buffer == NULL) {
         common_hal_audiodelays_echo_deinit(self);
@@ -285,8 +285,8 @@ audioio_get_buffer_result_t audiodelays_echo_get_buffer(audiodelays_echo_obj_t *
     }
 
     // get the effect values we need from the BlockInput. These may change at run time so you need to do bounds checking if required
-    mp_float_t mix = MIN(1.0, MAX(synthio_block_slot_get(&self->mix), 0.0));
-    mp_float_t decay = MIN(1.0, MAX(synthio_block_slot_get(&self->decay), 0.0));
+    mp_float_t mix = synthio_block_slot_get_limited(&self->mix, MICROPY_FLOAT_CONST(0.0), MICROPY_FLOAT_CONST(1.0));
+    mp_float_t decay = synthio_block_slot_get_limited(&self->decay, MICROPY_FLOAT_CONST(0.0), MICROPY_FLOAT_CONST(1.0));
 
     uint32_t delay_ms = (uint32_t)synthio_block_slot_get(&self->delay_ms);
     if (self->current_delay_ms != delay_ms) {
@@ -336,7 +336,7 @@ audioio_get_buffer_result_t audiodelays_echo_get_buffer(audiodelays_echo_obj_t *
 
         // If we have no sample keep the echo echoing
         if (self->sample == NULL) {
-            if (mix <= 0.01) {  // Mix of 0 is pure sample sound. We have no sample so no sound
+            if (mix <= MICROPY_FLOAT_CONST(0.01)) {  // Mix of 0 is pure sample sound. We have no sample so no sound
                 if (self->samples_signed) {
                     memset(word_buffer, 0, length * (self->bits_per_sample / 8));
                 } else {
@@ -406,7 +406,7 @@ audioio_get_buffer_result_t audiodelays_echo_get_buffer(audiodelays_echo_obj_t *
             int16_t *sample_src = (int16_t *)self->sample_remaining_buffer; // for 16-bit samples
             int8_t *sample_hsrc = (int8_t *)self->sample_remaining_buffer; // for 8-bit samples
 
-            if (mix <= 0.01) { // if mix is zero pure sample only
+            if (mix <= MICROPY_FLOAT_CONST(0.01)) { // if mix is zero pure sample only
                 for (uint32_t i = 0; i < n; i++) {
                     if (MP_LIKELY(self->bits_per_sample == 16)) {
                         word_buffer[i] = sample_src[i];
@@ -467,12 +467,12 @@ audioio_get_buffer_result_t audiodelays_echo_get_buffer(audiodelays_echo_obj_t *
                     word = echo + sample_word;
 
                     if (MP_LIKELY(self->bits_per_sample == 16)) {
-                        word_buffer[i] = (int16_t)((sample_word * (1.0 - mix)) + (word * mix));
+                        word_buffer[i] = (int16_t)((sample_word * (MICROPY_FLOAT_CONST(1.0) - mix)) + (word * mix));
                         if (!self->samples_signed) {
                             word_buffer[i] ^= 0x8000;
                         }
                     } else {
-                        int8_t mixed = (int16_t)((sample_word * (1.0 - mix)) + (word * mix));
+                        int8_t mixed = (int16_t)((sample_word * (MICROPY_FLOAT_CONST(1.0) - mix)) + (word * mix));
                         if (self->samples_signed) {
                             hword_buffer[i] = mixed;
                         } else {
diff --git a/shared-module/audiofilters/Distortion.c b/shared-module/audiofilters/Distortion.c
@@ -204,10 +204,10 @@ audioio_get_buffer_result_t audiofilters_distortion_get_buffer(audiofilters_dist
     }
 
     // get the effect values we need from the BlockInput. These may change at run time so you need to do bounds checking if required
-    mp_float_t drive = synthio_block_slot_get_limited(&self->drive, 0.0, 1.0);
-    mp_float_t pre_gain = db_to_linear(synthio_block_slot_get_limited(&self->pre_gain, -60.0, 60.0));
-    mp_float_t post_gain = db_to_linear(synthio_block_slot_get_limited(&self->post_gain, -80.0, 24.0));
-    mp_float_t mix = synthio_block_slot_get_limited(&self->mix, 0.0, 1.0);
+    mp_float_t drive = synthio_block_slot_get_limited(&self->drive, MICROPY_FLOAT_CONST(0.0), MICROPY_FLOAT_CONST(1.0));
+    mp_float_t pre_gain = db_to_linear(synthio_block_slot_get_limited(&self->pre_gain, MICROPY_FLOAT_CONST(-60.0), MICROPY_FLOAT_CONST(60.0)));
+    mp_float_t post_gain = db_to_linear(synthio_block_slot_get_limited(&self->post_gain, MICROPY_FLOAT_CONST(-80.0), MICROPY_FLOAT_CONST(24.0)));
+    mp_float_t mix = synthio_block_slot_get_limited(&self->mix, MICROPY_FLOAT_CONST(0.0), MICROPY_FLOAT_CONST(1.0));
 
     // Switch our buffers to the other buffer
     self->last_buf_idx = !self->last_buf_idx;
diff --git a/shared-module/audiofilters/Filter.c b/shared-module/audiofilters/Filter.c
@@ -53,10 +53,6 @@ void common_hal_audiofilters_filter_construct(audiofilters_filter_obj_t *self,
     }
     common_hal_audiofilters_filter_set_filter(self, filter);
 
-    // If we did not receive a BlockInput we need to create a default float value
-    if (mix == MP_OBJ_NULL) {
-        mix = mp_obj_new_float(1.0);
-    }
     synthio_block_assign_slot(mix, &self->mix, MP_QSTR_mix);
 }
 
@@ -251,7 +247,7 @@ audioio_get_buffer_result_t audiofilters_filter_get_buffer(audiofilters_filter_o
     }
 
     // get the effect values we need from the BlockInput. These may change at run time so you need to do bounds checking if required
-    mp_float_t mix = MIN(1.0, MAX(synthio_block_slot_get(&self->mix), 0.0));
+    mp_float_t mix = synthio_block_slot_get_limited(&self->mix, MICROPY_FLOAT_CONST(0.0), MICROPY_FLOAT_CONST(1.0));
 
     // Switch our buffers to the other buffer
     self->last_buf_idx = !self->last_buf_idx;
@@ -305,7 +301,7 @@ audioio_get_buffer_result_t audiofilters_filter_get_buffer(audiofilters_filter_o
             int16_t *sample_src = (int16_t *)self->sample_remaining_buffer; // for 16-bit samples
             int8_t *sample_hsrc = (int8_t *)self->sample_remaining_buffer; // for 8-bit samples
 
-            if (mix <= 0.01 || !self->filter_states) { // if mix is zero pure sample only or no biquad filter objects are provided
+            if (mix <= MICROPY_FLOAT_CONST(0.01) || !self->filter_states) { // if mix is zero pure sample only or no biquad filter objects are provided
                 for (uint32_t i = 0; i < n; i++) {
                     if (MP_LIKELY(self->bits_per_sample == 16)) {
                         word_buffer[i] = sample_src[i];
@@ -346,7 +342,7 @@ audioio_get_buffer_result_t audiofilters_filter_get_buffer(audiofilters_filter_o
                             }
                         } else {
                             if (self->samples_signed) {
-                                hword_buffer[i + j] = (int8_t)((sample_hsrc[i + j] * (1.0 - mix)) + (self->filter_buffer[j] * mix));
+                                hword_buffer[i + j] = (int8_t)((sample_hsrc[i + j] * (MICROPY_FLOAT_CONST(1.0) - mix)) + (self->filter_buffer[j] * mix));
                             } else {
                                 hword_buffer[i + j] = (uint8_t)(((int8_t)(((uint8_t)sample_hsrc[i + j]) ^ 0x80) * (MICROPY_FLOAT_CONST(1.0) - mix)) + (self->filter_buffer[j] * mix)) ^ 0x80;
                             }
