audio: reduce code size

By placing certain fields in a fixed location in all sample types,
code can be reduced & reused. For instance, the same property object
can be used for every sample type's `sample_rate` property.

The sample proto functions like `sample_rate` become superfluous since
once an object is verified to support the audiosample protocol, direct
access to the fields in the base object is possible.

Author: jepler

ports/atmel-samd/audio_dma.c

@@ -251,7 +251,7 @@ audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
     }
 
 
-    if (audiosample_bits_per_sample(sample) == 16) {
+    if (audiosample_get_bits_per_sample(sample) == 16) {
         dma->beat_size = 2;
         dma->bytes_per_sample = 2;
     } else {
@@ -262,7 +262,7 @@ audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
         }
     }
     // Transfer both channels at once.
-    if (!single_channel_output && audiosample_channel_count(sample) == 2) {
+    if (!single_channel_output && audiosample_get_channel_count(sample) == 2) {
         dma->beat_size *= 2;
     }
 

ports/atmel-samd/audio_dma.h

@@ -10,6 +10,7 @@
 #include "py/obj.h"
 #include "shared-module/audiocore/RawSample.h"
 #include "shared-module/audiocore/WaveFile.h"
+#include "shared-module/audiocore/__init__.h"
 #include "supervisor/background_callback.h"
 
 typedef struct {
@@ -40,10 +41,6 @@ typedef enum {
     AUDIO_DMA_MEMORY_ERROR,
 } audio_dma_result;
 
-uint32_t audiosample_sample_rate(mp_obj_t sample_obj);
-uint8_t audiosample_bits_per_sample(mp_obj_t sample_obj);
-uint8_t audiosample_channel_count(mp_obj_t sample_obj);
-
 void audio_dma_init(audio_dma_t *dma);
 void audio_dma_reset(void);
 

ports/atmel-samd/common-hal/audiobusio/I2SOut.c

@@ -216,10 +216,10 @@ void common_hal_audiobusio_i2sout_play(audiobusio_i2sout_obj_t *self,
         mp_raise_RuntimeError(MP_ERROR_TEXT("Clock unit in use"));
     }
     #endif
-    uint8_t bits_per_sample = audiosample_bits_per_sample(sample);
+    uint8_t bits_per_sample = audiosample_get_bits_per_sample(sample);
     // We always output stereo so output twice as many bits.
     uint16_t bits_per_sample_output = bits_per_sample * 2;
-    uint16_t divisor = 48000000 / (bits_per_sample_output * audiosample_sample_rate(sample));
+    uint16_t divisor = 48000000 / (bits_per_sample_output * audiosample_get_sample_rate(sample));
     // Find a free GCLK to generate the MCLK signal.
     uint8_t gclk = find_free_gclk(divisor);
     if (gclk > GCLK_GEN_NUM) {
@@ -235,7 +235,7 @@ void common_hal_audiobusio_i2sout_play(audiobusio_i2sout_obj_t *self,
     } else {
         clkctrl |= I2S_CLKCTRL_FSOUTINV | I2S_CLKCTRL_BITDELAY_I2S;
     }
-    uint8_t channel_count = audiosample_channel_count(sample);
+    uint8_t channel_count = audiosample_get_channel_count(sample);
     if (channel_count > 2) {
         mp_raise_ValueError(MP_ERROR_TEXT("Too many channels in sample"));
     }
@@ -245,7 +245,7 @@ void common_hal_audiobusio_i2sout_play(audiobusio_i2sout_obj_t *self,
     #ifdef SAM_D5X_E5X
     uint32_t serctrl = (self->clock_unit << I2S_RXCTRL_CLKSEL_Pos) | I2S_TXCTRL_TXSAME_SAME;
     #endif
-    if (audiosample_channel_count(sample) == 1) {
+    if (audiosample_get_channel_count(sample) == 1) {
         serctrl |= SERCTRL(MONO_MONO);
     } else {
         serctrl |= SERCTRL(MONO_STEREO);

ports/atmel-samd/common-hal/audioio/AudioOut.c

@@ -333,7 +333,7 @@ void common_hal_audioio_audioout_play(audioio_audioout_obj_t *self,
         common_hal_audioio_audioout_stop(self);
     }
     audio_dma_result result = AUDIO_DMA_OK;
-    uint32_t sample_rate = audiosample_sample_rate(sample);
+    uint32_t sample_rate = audiosample_get_sample_rate(sample);
     #ifdef SAMD21
     const uint32_t max_sample_rate = 350000;
     #endif
@@ -364,12 +364,12 @@ void common_hal_audioio_audioout_play(audioio_audioout_obj_t *self,
         right_channel_reg = (uint32_t)&DAC->DATABUF[0].reg;
     }
 
-    size_t num_channels = audiosample_channel_count(sample);
+    size_t num_channels = audiosample_get_channel_count(sample);
 
     if (num_channels == 2 &&
         // Are DAC channels sequential?
         left_channel_reg + 2 == right_channel_reg &&
-        audiosample_bits_per_sample(sample) == 16) {
+        audiosample_get_bits_per_sample(sample) == 16) {
         result = audio_dma_setup_playback(&self->left_dma, sample, loop, false, 0,
             false /* output unsigned */,
             left_channel_reg,
@@ -403,7 +403,7 @@ void common_hal_audioio_audioout_play(audioio_audioout_obj_t *self,
         }
     }
     Tc *timer = tc_insts[self->tc_index];
-    set_timer_frequency(timer, audiosample_sample_rate(sample));
+    set_timer_frequency(timer, audiosample_get_sample_rate(sample));
     timer->COUNT16.CTRLBSET.reg = TC_CTRLBSET_CMD_RETRIGGER;
     while (timer->COUNT16.STATUS.bit.STOP == 1) {
     }

ports/espressif/common-hal/audiobusio/__init__.c

@@ -149,8 +149,8 @@ void port_i2s_play(i2s_t *self, mp_obj_t sample, bool loop) {
     port_i2s_pause(self);
     self->sample = sample;
     self->loop = loop;
-    self->bytes_per_sample = audiosample_bits_per_sample(sample) / 8;
-    self->channel_count = audiosample_channel_count(sample);
+    self->bytes_per_sample = audiosample_get_bits_per_sample(sample) / 8;
+    self->channel_count = audiosample_get_channel_count(sample);
     bool single_buffer;
     bool samples_signed;
     uint32_t max_buffer_length;
@@ -164,7 +164,7 @@ void port_i2s_play(i2s_t *self, mp_obj_t sample, bool loop) {
 
     audiosample_reset_buffer(self->sample, false, 0);
 
-    uint32_t sample_rate = audiosample_sample_rate(sample);
+    uint32_t sample_rate = audiosample_get_sample_rate(sample);
     i2s_std_clk_config_t clk_config = I2S_STD_CLK_DEFAULT_CONFIG(sample_rate);
     CHECK_ESP_RESULT(i2s_channel_reconfig_std_clock(self->handle, &clk_config));
 

ports/espressif/common-hal/audioio/AudioOut.c

@@ -595,16 +595,16 @@ void common_hal_audioio_audioout_play(audioio_audioout_obj_t *self,
 
     self->sample = sample;
     self->looping = loop;
-    freq_hz = audiosample_sample_rate(self->sample);
+    freq_hz = audiosample_get_sample_rate(self->sample);
 
     if (freq_hz != self->freq_hz) {
         common_hal_audioio_audioout_deinit(self);
         self->freq_hz = freq_hz;
         audioout_init(self);
     }
 
-    samples_size = audiosample_bits_per_sample(self->sample);
-    channel_count = audiosample_channel_count(self->sample);
+    samples_size = audiosample_get_bits_per_sample(self->sample);
+    channel_count = audiosample_get_channel_count(self->sample);
     audiosample_get_buffer_structure(self->sample, false,
         &_single_buffer, &samples_signed,
         &_max_buffer_length, &_spacing);

ports/mimxrt10xx/common-hal/audiobusio/__init__.c

@@ -374,11 +374,11 @@ static void set_sai_clocking_for_sample_rate(uint32_t sample_rate) {
 void port_i2s_play(i2s_t *self, mp_obj_t sample, bool loop) {
     self->sample = sample;
     self->loop = loop;
-    self->bytes_per_sample = audiosample_bits_per_sample(sample) / 8;
-    self->channel_count = audiosample_channel_count(sample);
+    self->bytes_per_sample = audiosample_get_bits_per_sample(sample) / 8;
+    self->channel_count = audiosample_get_channel_count(sample);
     int instance = SAI_GetInstance(self->peripheral);
     i2s_playing |= (1 << instance);
-    uint32_t sample_rate = audiosample_sample_rate(sample);
+    uint32_t sample_rate = audiosample_get_sample_rate(sample);
     if (sample_rate != self->sample_rate) {
         if (__builtin_popcount(i2s_playing) <= 1) {
             // as this is the first/only i2s instance playing audio, we can

ports/nordic/common-hal/audiobusio/I2SOut.c

@@ -240,8 +240,8 @@ void common_hal_audiobusio_i2sout_play(audiobusio_i2sout_obj_t *self,
 
     self->sample = sample;
     self->loop = loop;
-    uint32_t sample_rate = audiosample_sample_rate(sample);
-    self->bytes_per_sample = audiosample_bits_per_sample(sample) / 8;
+    uint32_t sample_rate = audiosample_get_sample_rate(sample);
+    self->bytes_per_sample = audiosample_get_bits_per_sample(sample) / 8;
 
     uint32_t max_buffer_length;
     bool single_buffer, samples_signed;

ports/nordic/common-hal/audiopwmio/PWMAudioOut.c

@@ -223,15 +223,15 @@ void common_hal_audiopwmio_pwmaudioout_play(audiopwmio_pwmaudioout_obj_t *self,
     self->sample = sample;
     self->loop = loop;
 
-    uint32_t sample_rate = audiosample_sample_rate(sample);
-    self->bytes_per_sample = audiosample_bits_per_sample(sample) / 8;
+    uint32_t sample_rate = audiosample_get_sample_rate(sample);
+    self->bytes_per_sample = audiosample_get_bits_per_sample(sample) / 8;
 
     uint32_t max_buffer_length;
     uint8_t spacing;
     audiosample_get_buffer_structure(sample, /* single channel */ false,
         &self->single_buffer, &self->signed_to_unsigned, &max_buffer_length,
         &spacing);
-    self->sample_channel_count = audiosample_channel_count(sample);
+    self->sample_channel_count = audiosample_get_channel_count(sample);
 
     mp_arg_validate_length_max(max_buffer_length, UINT16_MAX, MP_QSTR_buffer);
 

ports/raspberrypi/audio_dma.c

@@ -203,7 +203,7 @@ audio_dma_result audio_dma_setup_playback(
     dma->output_signed = output_signed;
     dma->sample_spacing = 1;
     dma->output_resolution = output_resolution;
-    dma->sample_resolution = audiosample_bits_per_sample(sample);
+    dma->sample_resolution = audiosample_get_bits_per_sample(sample);
     dma->output_register_address = output_register_address;
     dma->swap_channel = swap_channel;
 
@@ -250,7 +250,7 @@ audio_dma_result audio_dma_setup_playback(
         dma->output_size = 1;
     }
     // Transfer both channels at once.
-    if (!single_channel_output && audiosample_channel_count(sample) == 2) {
+    if (!single_channel_output && audiosample_get_channel_count(sample) == 2) {
         dma->output_size *= 2;
     }
     enum dma_channel_transfer_size dma_size = DMA_SIZE_8;