Re-enable PDMIn without ASF and using the helpers added with

I2SOut.

The API is almost the same except the frequency attribute has been
renamed to sample_rate so that its less likely to be confused with
frequencies within the audio itself.

Fixes #263.

13 files changed, 373 insertions, 170 deletions

diff --git a/ports/atmel-samd/Makefile b/ports/atmel-samd/Makefile
index 6b661cd84..bb2d92b4a 100644
--- a/ports/atmel-samd/Makefile
+++ b/ports/atmel-samd/Makefile
@@ -299,13 +299,13 @@ SRC_COMMON_HAL = \
 	usb_hid/Device.c \
 	audioio/__init__.c \
 	audioio/AudioOut.c \
-#	audiobusio/PDMIn.c \
-	touchio/__init__.c \
+#	touchio/__init__.c \
 	touchio/TouchIn.c \
 
 ifeq ($(INTERNAL_LIBM),1)
 SRC_LIBM = $(addprefix lib/,\
 	libm/math.c \
+	libm/roundf.c \
 	libm/fmodf.c \
 	libm/nearbyintf.c \
 	libm/ef_sqrt.c \
@@ -363,16 +363,14 @@ SRC_SHARED_MODULE = \
 	uheap/__init__.c \
 	ustack/__init__.c
 
-ifeq ($(CHIP_FAMILY),samd21)
-SRC_COMMON_HAL += \
-	audiobusio/__init__.c \
-	audiobusio/I2SOut.c
-endif
+# The smallest SAMD51 packages don't have I2S. Everything else does.
 ifneq ($(CHIP_VARIANT),SAMD51G18A)
 	ifneq ($(CHIP_VARIANT),SAMD51G19A)
 		SRC_COMMON_HAL += \
 			audiobusio/__init__.c \
-			audiobusio/I2SOut.c
+			audiobusio/I2SOut.c \
+			audiobusio/PDMIn.c
+		SRC_C += i2s.c
 	endif
 endif
 
diff --git a/ports/atmel-samd/audio_dma.c b/ports/atmel-samd/audio_dma.c
index 7030009d9..227a2e188 100644
--- a/ports/atmel-samd/audio_dma.c
+++ b/ports/atmel-samd/audio_dma.c
@@ -267,7 +267,6 @@ audio_dma_result audio_dma_setup_playback(audio_dma_t* dma,
         turn_on_event_system();
         dma->event_channel = find_sync_event_channel();
         init_event_channel_interrupt(dma->event_channel, CORE_GCLK, EVSYS_ID_GEN_DMAC_CH_0 + dma_channel);
-        find_sync_event_channel();
 
         // We keep the audio_dma_t for internal use and the sample as a root pointer because it
         // contains the audiodma structure.
diff --git a/ports/atmel-samd/audio_dma.h b/ports/atmel-samd/audio_dma.h
index deecd27f3..fd2e6a153 100644
--- a/ports/atmel-samd/audio_dma.h
+++ b/ports/atmel-samd/audio_dma.h
@@ -64,6 +64,8 @@ uint8_t audiosample_channel_count(mp_obj_t sample_obj);
 void audio_dma_init(audio_dma_t* dma);
 void audio_dma_reset(void);
 
+uint8_t find_free_audio_dma_channel(void);
+
 // This sets everything up but doesn't start the timer.
 // Sample is the python object for the sample to play.
 // loop is true if we should loop the sample.
diff --git a/ports/atmel-samd/common-hal/audiobusio/I2SOut.c b/ports/atmel-samd/common-hal/audiobusio/I2SOut.c
index ec21636d0..65dae5dcb 100644
--- a/ports/atmel-samd/common-hal/audiobusio/I2SOut.c
+++ b/ports/atmel-samd/common-hal/audiobusio/I2SOut.c
@@ -48,7 +48,8 @@
 #include "audio_dma.h"
 #include "clocks.h"
 #include "events.h"
-#include "samd21_pins.h"
+#include "i2s.h"
+#include "pins.h"
 #include "shared_dma.h"
 #include "timers.h"
 
@@ -155,18 +156,7 @@ void common_hal_audiobusio_i2sout_construct(audiobusio_i2sout_obj_t* self,
     self->clock_unit = ws_clock_unit;
     self->serializer = serializer;
 
-    // Make sure the I2S peripheral is running so we can see if the resources we need are free.
-    #ifdef SAMD51
-    hri_mclk_set_APBDMASK_I2S_bit(MCLK);
-
-    // Connect the clock units to the 2mhz clock by default. They can't reset without it.
-    connect_gclk_to_peripheral(5, I2S_GCLK_ID_0);
-    connect_gclk_to_peripheral(5, I2S_GCLK_ID_1);
-    #endif
-
-    #ifdef SAMD21
-    _pm_enable_bus_clock(PM_BUS_APBC, I2S);
-    #endif
+    turn_on_i2s();
 
     if (I2S->CTRLA.bit.ENABLE == 0) {
         I2S->CTRLA.bit.SWRST = 1;
@@ -281,8 +271,7 @@ void common_hal_audiobusio_i2sout_play(audiobusio_i2sout_obj_t* self,
     }
 
     // Configure the I2S peripheral
-    I2S->CTRLA.bit.ENABLE = 0;
-    while (I2S->SYNCBUSY.bit.ENABLE == 1) {}
+    i2s_set_enable(false);
 
     I2S->CLKCTRL[self->clock_unit].reg = clkctrl;
     #ifdef SAMD21
@@ -296,8 +285,7 @@ void common_hal_audiobusio_i2sout_play(audiobusio_i2sout_obj_t* self,
     enable_clock_generator(self->gclk, CLOCK_48MHZ, divisor);
     connect_gclk_to_peripheral(self->gclk, I2S_GCLK_ID_0 + self->clock_unit);
 
-    I2S->CTRLA.bit.ENABLE = 1;
-    while (I2S->SYNCBUSY.bit.ENABLE == 1) {}
+    i2s_set_enable(true);
 
     #ifdef SAMD21
     uint32_t tx_register = (uint32_t) &I2S->DATA[self->serializer].reg;
diff --git a/ports/atmel-samd/common-hal/audiobusio/PDMIn.c b/ports/atmel-samd/common-hal/audiobusio/PDMIn.c
index e0ac3f9a3..7788335d1 100644
--- a/ports/atmel-samd/common-hal/audiobusio/PDMIn.c
+++ b/ports/atmel-samd/common-hal/audiobusio/PDMIn.c
@@ -37,17 +37,32 @@
 #include "shared-bindings/audiobusio/PDMIn.h"
 #include "shared-bindings/microcontroller/Pin.h"
 
-#include "samd21_pins.h"
-
+#include "atmel_start_pins.h"
+#include "hal/include/hal_gpio.h"
+#include "hal/utils/include/utils.h"
+
+#include "audio_dma.h"
+#include "clocks.h"
+#include "events.h"
+#include "i2s.h"
+#include "pins.h"
 #include "shared_dma.h"
 #include "tick.h"
 
 #define OVERSAMPLING 64
-#define SAMPLES_PER_BUFFER 32
+#define SAMPLES_PER_BUFFER 64
 
 // MEMS microphones must be clocked at at least 1MHz.
 #define MIN_MIC_CLOCK 1000000
 
+#ifdef SAMD21
+#define SERCTRL(name) I2S_SERCTRL_ ## name
+#endif
+
+#ifdef SAMD51
+#define SERCTRL(name) I2S_RXCTRL_ ## name
+#endif
+
 void pdmin_reset(void) {
     while (I2S->SYNCBUSY.reg & I2S_SYNCBUSY_ENABLE) {}
     I2S->INTENCLR.reg = I2S_INTENCLR_MASK;
@@ -60,20 +75,33 @@ void pdmin_reset(void) {
 void common_hal_audiobusio_pdmin_construct(audiobusio_pdmin_obj_t* self,
                                            const mcu_pin_obj_t* clock_pin,
                                            const mcu_pin_obj_t* data_pin,
-                                           uint32_t frequency,
+                                           uint32_t sample_rate,
                                            uint8_t bit_depth,
                                            bool mono,
                                            uint8_t oversample) {
     self->clock_pin = clock_pin; // PA10, PA20 -> SCK0, PB11 -> SCK1
-    if (clock_pin == &pin_PA10
-    #ifdef PIN_PA20
-        || clock_pin == &pin_PA20
+    #ifdef SAMD21
+        if (clock_pin == &pin_PA10
+        #ifdef PIN_PA20
+            || clock_pin == &pin_PA20
+        #endif
+            ) {
+            self->clock_unit = 0;
+        #ifdef PIN_PB11
+        } else if (clock_pin == &pin_PB11) {
+            self->clock_unit = 1;
+        #endif
     #endif
+    #ifdef SAMD51
+        if (clock_pin == &pin_PA10 || clock_pin == &pin_PB16) {
+            self->clock_unit = 0;
+    } else if (clock_pin == &pin_PB12
+        #ifdef PIN_PB28
+        || data_pin == &pin_PB28) {
+        #else
         ) {
-        self->clock_unit = 0;
-    #ifdef PIN_PB11
-    } else if (clock_pin == &pin_PB11) {
-        self->clock_unit = 1;
+        #endif
+            self->clock_unit = 1;
     #endif
     } else {
         mp_raise_ValueError("Invalid clock pin");
@@ -81,80 +109,108 @@ void common_hal_audiobusio_pdmin_construct(audiobusio_pdmin_obj_t* self,
 
     self->data_pin = data_pin; // PA07, PA19 -> SD0, PA08, PB16 -> SD1
 
+    #ifdef SAMD21
     if (data_pin == &pin_PA07 || data_pin == &pin_PA19) {
         self->serializer = 0;
     } else if (data_pin == &pin_PA08
-        #ifdef PB16
+        #ifdef PIN_PB16
         || data_pin == &pin_PB16) {
         #else
         ) {
         #endif
         self->serializer = 1;
+    #endif
+    #ifdef SAMD51
+    if (data_pin == &pin_PB10 || data_pin == &pin_PA22) {
+        self->serializer = 1;
+    #endif
     } else {
         mp_raise_ValueError("Invalid data pin");
     }
 
-    claim_pin(clock_pin);
-    claim_pin(data_pin);
-
-    if (MP_STATE_VM(audiodma_block_counter) == NULL &&
-        !allocate_block_counter()) {
-        mp_raise_RuntimeError("Unable to allocate audio DMA block counter.");
-    }
-
     if (!(bit_depth == 16 || bit_depth == 8) || !mono || oversample != OVERSAMPLING) {
         mp_raise_NotImplementedError("Only 8 or 16 bit mono with " MP_STRINGIFY(OVERSAMPLING) "x oversampling is supported.");
     }
 
-    // TODO(tannewt): Use the DPLL to get a more precise sampling rate.
-    // DFLL -> GCLK (/600 for 8khz, /300 for 16khz and /150 for 32khz) -> DPLL (*(63 + 1)) -> GCLK ( / 10) -> 512khz
+    turn_on_i2s();
 
-    i2s_init(&self->i2s_instance, I2S);
-    struct i2s_clock_unit_config config_clock_unit;
-    i2s_clock_unit_get_config_defaults(&config_clock_unit);
-    config_clock_unit.clock.gclk_src = GCLK_GENERATOR_3;
+    if (I2S->CTRLA.bit.ENABLE == 0) {
+        I2S->CTRLA.bit.SWRST = 1;
+        while (I2S->CTRLA.bit.SWRST == 1) {}
+    } else {
+        #ifdef SAMD21
+        if ((I2S->CTRLA.vec.SEREN & (1 << self->serializer)) != 0) {
+            mp_raise_RuntimeError("Serializer in use");
+        }
+        #endif
+        #ifdef SAMD51
+        if (I2S->CTRLA.bit.RXEN == 1) {
+            mp_raise_RuntimeError("Serializer in use");
+        }
+        #endif
+    }
+    #ifdef SAMD51
+    #define GPIO_I2S_FUNCTION GPIO_PIN_FUNCTION_J
+    #endif
+    #ifdef SAMD21
+    #define GPIO_I2S_FUNCTION GPIO_PIN_FUNCTION_G
+    #endif
+    assert_pin_free(clock_pin);
+    assert_pin_free(data_pin);
+
+    uint32_t clock_divisor = (uint32_t) roundf( 48000000.0f / sample_rate / oversample);
+    float mic_clock_freq = 48000000.0f / clock_divisor;
+    self->sample_rate =  mic_clock_freq / oversample;
+    if (mic_clock_freq <  MIN_MIC_CLOCK || clock_divisor == 0) {
+        mp_raise_ValueError("sampling rate out of range");
+    }
+    // Find a free GCLK to generate the MCLK signal.
+    uint8_t gclk = find_free_gclk(clock_divisor);
+    if (gclk > GCLK_GEN_NUM) {
+        mp_raise_RuntimeError("Unable to find free GCLK");
+    }
+    self->gclk = gclk;
 
-    config_clock_unit.clock.mck_src = I2S_MASTER_CLOCK_SOURCE_GCLK;
-    config_clock_unit.clock.mck_out_enable = false;
+    enable_clock_generator(self->gclk, CLOCK_48MHZ, clock_divisor);
+    connect_gclk_to_peripheral(self->gclk, I2S_GCLK_ID_0 + self->clock_unit);
 
-    config_clock_unit.clock.sck_src = I2S_SERIAL_CLOCK_SOURCE_MCKDIV;
-    uint32_t clock_divisor = (uint32_t) roundf( 8000000.0f / frequency / oversample);
-    config_clock_unit.clock.sck_div = clock_divisor;
-    float mic_clock_freq = 8000000.0f / clock_divisor;
-    self->frequency =  mic_clock_freq / oversample;
-    if (mic_clock_freq <  MIN_MIC_CLOCK || clock_divisor == 0 || clock_divisor > 255) {
-        mp_raise_ValueError("sampling frequency out of range");
-    }
+    // Clock unit configuration
+
+    uint32_t clkctrl = I2S_CLKCTRL_MCKSEL_GCLK |
+                       I2S_CLKCTRL_NBSLOTS(2) |
+                       I2S_CLKCTRL_FSWIDTH_SLOT |
+                       I2S_CLKCTRL_SLOTSIZE_16;
+
+    // Serializer configuration
+    #ifdef SAMD21
+    uint32_t serctrl = (self->clock_unit << I2S_SERCTRL_CLKSEL_Pos) | SERCTRL(SERMODE_PDM2) | SERCTRL(DATASIZE_32);
+    #endif
+    #ifdef SAMD51
+    uint32_t serctrl = (self->clock_unit << I2S_RXCTRL_CLKSEL_Pos) | SERCTRL(SERMODE_PDM2) | SERCTRL(DATASIZE_32);
+    #endif
 
-    config_clock_unit.frame.number_slots = 2;
-    config_clock_unit.frame.slot_size = I2S_SLOT_SIZE_16_BIT;
-    config_clock_unit.frame.data_delay = I2S_DATA_DELAY_0;
-
-    config_clock_unit.frame.frame_sync.width = I2S_FRAME_SYNC_WIDTH_SLOT;
-
-    config_clock_unit.mck_pin.enable = false;
-    config_clock_unit.sck_pin.enable = true;
-    config_clock_unit.sck_pin.gpio = self->clock_pin->pin;
-    // Mux is always the same.
-    config_clock_unit.sck_pin.mux = 6L;
-    config_clock_unit.fs_pin.enable = false;
-    i2s_clock_unit_set_config(&self->i2s_instance, self->clock_unit, &config_clock_unit);
-
-    struct i2s_serializer_config config_serializer;
-    i2s_serializer_get_config_defaults(&config_serializer);
-    config_serializer.clock_unit = self->clock_unit;
-    config_serializer.mode = I2S_SERIALIZER_PDM2;
-    config_serializer.data_size = I2S_DATA_SIZE_32BIT;
-    config_serializer.data_pin.gpio = self->data_pin->pin;
-    // Mux is always the same.
-    config_serializer.data_pin.mux = 6L;
-    config_serializer.data_pin.enable = true;
-    i2s_serializer_set_config(&self->i2s_instance, self->serializer, &config_serializer);
-    i2s_enable(&self->i2s_instance);
+    // Configure the I2S peripheral
+    i2s_set_enable(false);
+
+    I2S->CLKCTRL[self->clock_unit].reg = clkctrl;
+    #ifdef SAMD21
+    I2S->SERCTRL[self->serializer].reg = serctrl;
+    #endif
+    #ifdef SAMD51
+    I2S->RXCTRL.reg = serctrl;
+    #endif
+
+    i2s_set_enable(true);
 
     // Run the serializer all the time. This eliminates startup delay for the microphone.
-    i2s_clock_unit_enable(&self->i2s_instance, self->clock_unit);
-    i2s_serializer_enable(&self->i2s_instance, self->serializer);
+    i2s_set_clock_unit_enable(self->clock_unit, true);
+    i2s_set_serializer_enable(self->serializer, true);
+
+    claim_pin(clock_pin);
+    claim_pin(data_pin);
+
+    gpio_set_pin_function(self->clock_pin->pin, GPIO_I2S_FUNCTION);
+    gpio_set_pin_function(self->data_pin->pin, GPIO_I2S_FUNCTION);
 
     self->bytes_per_sample = oversample >> 3;
     self->bit_depth = bit_depth;
@@ -168,10 +224,15 @@ void common_hal_audiobusio_pdmin_deinit(audiobusio_pdmin_obj_t* self) {
     if (common_hal_audiobusio_pdmin_deinited(self)) {
         return;
     }
-    i2s_disable(&self->i2s_instance);
-    i2s_serializer_disable(&self->i2s_instance, self->serializer);
-    i2s_clock_unit_disable(&self->i2s_instance, self->clock_unit);
-    i2s_reset(&self->i2s_instance);
+
+    i2s_set_serializer_enable(self->serializer, false);
+    i2s_set_clock_unit_enable(self->clock_unit, false);
+
+    i2s_set_enable(false);
+
+    disconnect_gclk_from_peripheral(self->gclk, I2S_GCLK_ID_0 + self->clock_unit);
+    disable_clock_generator(self->gclk);
+
     reset_pin(self->clock_pin->pin);
     reset_pin(self->data_pin->pin);
     self->clock_pin = mp_const_none;
@@ -182,65 +243,67 @@ uint8_t common_hal_audiobusio_pdmin_get_bit_depth(audiobusio_pdmin_obj_t* self)
     return self->bit_depth;
 }
 
-uint32_t common_hal_audiobusio_pdmin_get_frequency(audiobusio_pdmin_obj_t* self) {
-    return self->frequency;
+uint32_t common_hal_audiobusio_pdmin_get_sample_rate(audiobusio_pdmin_obj_t* self) {
+    return self->sample_rate;
 }
 
 static void setup_dma(audiobusio_pdmin_obj_t* self, uint32_t length,
-        DmacDescriptor* second_descriptor,
-        uint8_t words_per_buffer, uint8_t words_per_sample,
-        uint32_t* first_buffer, uint32_t* second_buffer) {
-    // Set up the DMA
-    struct dma_descriptor_config descriptor_config;
-    dma_descriptor_get_config_defaults(&descriptor_config);
-    descriptor_config.beat_size = DMA_BEAT_SIZE_WORD;
-    descriptor_config.step_selection = DMA_STEPSEL_SRC;
-    descriptor_config.source_address = (uint32_t)&I2S->DATA[self->serializer];
-    descriptor_config.src_increment_enable = false;
+                      DmacDescriptor* descriptor,
+                      DmacDescriptor* second_descriptor,
+                      uint32_t words_per_buffer, uint8_t words_per_sample,
+                      uint32_t* first_buffer, uint32_t* second_buffer) {
+    descriptor->BTCTRL.reg = DMAC_BTCTRL_VALID |
+                             DMAC_BTCTRL_BLOCKACT_NOACT |
+                             DMAC_BTCTRL_EVOSEL_BLOCK |
+                             DMAC_BTCTRL_DSTINC |
+                             DMAC_BTCTRL_BEATSIZE_WORD;
+
     // Block transfer count is the number of beats per block (aka descriptor).
     // In this case there are two bytes per beat so divide the length by two.
     uint16_t block_transfer_count = words_per_buffer;
     if (length * words_per_sample < words_per_buffer) {
         block_transfer_count = length * words_per_sample;
     }
-    descriptor_config.block_transfer_count = block_transfer_count;
-    descriptor_config.destination_address = ((uint32_t) first_buffer + sizeof(uint32_t) * block_transfer_count);
-    descriptor_config.event_output_selection = DMA_EVENT_OUTPUT_BLOCK;
-    descriptor_config.next_descriptor_address = 0;
+
+    descriptor->BTCNT.reg = block_transfer_count;
+    descriptor->DSTADDR.reg = ((uint32_t) first_buffer + sizeof(uint32_t) * block_transfer_count);
+    descriptor->DESCADDR.reg = 0;
     if (length * words_per_sample > words_per_buffer) {
-        descriptor_config.next_descriptor_address = ((uint32_t)second_descriptor);
+        descriptor->DESCADDR.reg = ((uint32_t)second_descriptor);
     }
-    dma_descriptor_create(audio_dma.descriptor, &descriptor_config);
+    #ifdef SAMD21
+    descriptor->SRCADDR.reg = (uint32_t)&I2S->DATA[self->serializer];
+    #endif
+    #ifdef SAMD51
+    descriptor->SRCADDR.reg = (uint32_t)&I2S->RXDATA;
+    #endif
 
     // Do we need more values than will fit in the first buffer?
     // If so, set up a second buffer chained to be filled after the first buffer.
     if (length * words_per_sample > words_per_buffer) {
         block_transfer_count = words_per_buffer;
-        descriptor_config.next_descriptor_address = ((uint32_t)audio_dma.descriptor);
+        second_descriptor->DESCADDR.reg = ((uint32_t)descriptor);
         if (length * words_per_sample < 2 * words_per_buffer) {
             // Length needed is more than one buffer but less than two.
             // Subtract off the size of the first buffer, and what remains is the count we need.
             block_transfer_count = length * words_per_sample - words_per_buffer;
-            descriptor_config.next_descriptor_address = 0;
+            second_descriptor->DESCADDR.reg = 0;
         }
-        descriptor_config.block_transfer_count = block_transfer_count;
-        descriptor_config.destination_address = ((uint32_t) second_buffer + sizeof(uint32_t) * block_transfer_count);
-        dma_descriptor_create(second_descriptor, &descriptor_config);
-    }
-
-    switch_audiodma_trigger(I2S_DMAC_ID_RX_0 + self->serializer);
-}
+        second_descriptor->DSTADDR.reg = ((uint32_t) second_buffer + sizeof(uint32_t) * block_transfer_count);
 
-void start_dma(audiobusio_pdmin_obj_t* self) {
-    dma_start_transfer_job(&audio_dma);
-    tc_start_counter(MP_STATE_VM(audiodma_block_counter));
-    I2S->DATA[1].reg = I2S->DATA[1].reg;
-}
-
-void stop_dma(audiobusio_pdmin_obj_t* self) {
-    // Shutdown the DMA: serializer keeps running.
-    tc_stop_counter(MP_STATE_VM(audiodma_block_counter));
-    dma_abort_job(&audio_dma);
+        second_descriptor->BTCNT.reg = block_transfer_count;
+        #ifdef SAMD21
+        second_descriptor->SRCADDR.reg = (uint32_t)&I2S->DATA[self->serializer];
+        #endif
+        #ifdef SAMD51
+        second_descriptor->SRCADDR.reg = (uint32_t)&I2S->RXDATA;
+        #endif
+        second_descriptor->BTCTRL.reg = DMAC_BTCTRL_VALID |
+                                        DMAC_BTCTRL_BLOCKACT_NOACT |
+                                        DMAC_BTCTRL_EVOSEL_BLOCK |
+                                        DMAC_BTCTRL_DSTINC |
+                                        DMAC_BTCTRL_BEATSIZE_WORD;
+    }
 }
 
 // a windowed sinc filter for 44 khz, 64 samples
@@ -290,20 +353,27 @@ static uint16_t filter_sample(uint32_t pdm_samples[4]) {
 // output_buffer_length is the number of slots, not the number of bytes.
 uint32_t common_hal_audiobusio_pdmin_record_to_buffer(audiobusio_pdmin_obj_t* self,
         uint16_t* output_buffer, uint32_t output_buffer_length) {
+    uint8_t dma_channel = find_free_audio_dma_channel();
+    uint8_t event_channel = find_sync_event_channel();
+
     // We allocate two buffers on the stack to use for double buffering.
     const uint8_t samples_per_buffer = SAMPLES_PER_BUFFER;
     // For every word we record, we throw away 2 bytes of a phantom second channel.
-    const uint8_t words_per_sample = self->bytes_per_sample / 2;
-    const uint8_t words_per_buffer = samples_per_buffer * words_per_sample;
+    uint8_t words_per_sample = self->bytes_per_sample / 2;
+    uint32_t words_per_buffer = samples_per_buffer * words_per_sample;
     uint32_t first_buffer[words_per_buffer];
     uint32_t second_buffer[words_per_buffer];
 
+    turn_on_event_system();
+
     COMPILER_ALIGNED(16) DmacDescriptor second_descriptor;
 
-    setup_dma(self, output_buffer_length, &second_descriptor, words_per_buffer,
-       words_per_sample, first_buffer, second_buffer);
+    setup_dma(self, output_buffer_length, dma_descriptor(dma_channel), &second_descriptor,
+              words_per_buffer, words_per_sample, first_buffer, second_buffer);
 
-    start_dma(self);
+    dma_configure(dma_channel, I2S_DMAC_ID_RX_0, true);
+    init_event_channel_interrupt(event_channel, CORE_GCLK, EVSYS_ID_GEN_DMAC_CH_0 + dma_channel);
+    dma_enable_channel(dma_channel);
 
     // Record
     uint32_t buffers_processed = 0;
@@ -311,24 +381,23 @@ uint32_t common_hal_audiobusio_pdmin_record_to_buffer(audiobusio_pdmin_obj_t* se
 
     uint32_t remaining_samples_needed = output_buffer_length;
     while (values_output < output_buffer_length) {
+        if (event_interrupt_overflow(event_channel)) {
+            // Looks like we aren't keeping up. We shouldn't skip a buffer so stop early.
+            break;
+        }
         // Wait for the next buffer to fill
-        uint32_t block_counter;
-        while ((block_counter = tc_get_count_value(MP_STATE_VM(audiodma_block_counter))) == buffers_processed) {
+        while (!event_interrupt_active(event_channel)) {
             #ifdef MICROPY_VM_HOOK_LOOP
                 MICROPY_VM_HOOK_LOOP
             #endif
         }
-        if (block_counter != (buffers_processed + 1)) {
-            // Looks like we aren't keeping up. We shouldn't skip a buffer.
-            break;
-        }
 
         // The mic is running all the time, so we don't need to wait the usual 10msec or 100msec
         // for it to start up.
 
         // Flip back and forth between processing the first and second buffers.
         uint32_t *buffer = first_buffer;
-        DmacDescriptor* descriptor = audio_dma.descriptor;
+        DmacDescriptor* descriptor = dma_descriptor(dma_channel);
         if (buffers_processed % 2 == 1) {
             buffer = second_buffer;
             descriptor = &second_descriptor;
@@ -375,7 +444,8 @@ uint32_t common_hal_audiobusio_pdmin_record_to_buffer(audiobusio_pdmin_obj_t* se
         }
     }
 
-    stop_dma(self);
+    disable_event_channel(event_channel);
+    dma_disable_channel(dma_channel);
 
     return values_output;
 }
diff --git a/ports/atmel-samd/common-hal/audiobusio/PDMIn.h b/ports/atmel-samd/common-hal/audiobusio/PDMIn.h
index 50dc60262..1156c6421 100644
--- a/ports/atmel-samd/common-hal/audiobusio/PDMIn.h
+++ b/ports/atmel-samd/common-hal/audiobusio/PDMIn.h
@@ -36,11 +36,12 @@ typedef struct {
     mp_obj_base_t base;
     const mcu_pin_obj_t *clock_pin;
     const mcu_pin_obj_t *data_pin;
-    uint32_t frequency;
+    uint32_t sample_rate;
     uint8_t serializer;
     uint8_t clock_unit;
     uint8_t bytes_per_sample;
     uint8_t bit_depth;
+    uint8_t gclk;
 } audiobusio_pdmin_obj_t;
 
 void pdmin_reset(void);
diff --git a/ports/atmel-samd/events.c b/ports/atmel-samd/events.c
index 1854692ae..05a2ae8a7 100644
--- a/ports/atmel-samd/events.c
+++ b/ports/atmel-samd/events.c
@@ -91,7 +91,7 @@ uint8_t find_async_event_channel(void) {
 #define EVSYS_SYNCH_NUM EVSYS_CHANNELS
 #endif
 uint8_t find_sync_event_channel(void) {
-    int8_t channel;
+    uint8_t channel;
     for (channel = 0; channel < EVSYS_SYNCH_NUM; channel++) {
         if (channel_free(channel)) {
             break;
@@ -145,6 +145,8 @@ void init_event_channel_interrupt(uint8_t channel, uint8_t gclk, uint8_t generat
     EVSYS->Channel[channel].CHANNEL.reg = EVSYS_CHANNEL_EVGEN(generator) |
                                           EVSYS_CHANNEL_PATH_SYNCHRONOUS |
                                           EVSYS_CHANNEL_EDGSEL_RISING_EDGE;
+    EVSYS->Channel[channel].CHINTFLAG.reg = EVSYS_CHINTFLAG_EVD | EVSYS_CHINTFLAG_OVR;
+    EVSYS->Channel[channel].CHINTENSET.reg = EVSYS_CHINTENSET_EVD | EVSYS_CHINTENSET_OVR;
     #endif
     #ifdef SAMD21
     EVSYS->CHANNEL.reg = EVSYS_CHANNEL_CHANNEL(channel) |
@@ -187,8 +189,25 @@ bool event_interrupt_active(uint8_t channel) {
     // Only clear if we know its active, otherwise there is the possibility it becomes after we
     // check but before we clear.
     if (active) {
-        EVSYS->Channel[channel].CHINTFLAG.reg = EVSYS_CHINTFLAG_EVD;
+        EVSYS->Channel[channel].CHINTFLAG.reg = EVSYS_CHINTFLAG_EVD | EVSYS_CHINTFLAG_OVR;
     }
     #endif
     return active;
 }
+
+bool event_interrupt_overflow(uint8_t channel) {
+    bool overflow = false;
+    #ifdef SAMD21
+    if (channel > 7) {
+        uint8_t value = 1 << (channel - 7);
+        overflow = (EVSYS->INTFLAG.reg & EVSYS_INTFLAG_OVRp8(value)) != 0;
+    } else {
+        uint8_t value = 1 << channel;
+        overflow = (EVSYS->INTFLAG.reg & EVSYS_INTFLAG_OVR(value)) != 0;
+    }
+    #endif
+    #ifdef SAMD51
+    overflow = EVSYS->Channel[channel].CHINTFLAG.bit.OVR;
+    #endif
+    return overflow;
+}
diff --git a/ports/atmel-samd/events.h b/ports/atmel-samd/events.h
index 69215c3e7..64093c0ab 100644
--- a/ports/atmel-samd/events.h
+++ b/ports/atmel-samd/events.h
@@ -42,5 +42,6 @@ void connect_event_user_to_channel(uint8_t user, uint8_t channel);
 void init_async_event_channel(uint8_t channel, uint8_t generator);
 void init_event_channel_interrupt(uint8_t channel, uint8_t gclk, uint8_t generator);
 bool event_interrupt_active(uint8_t channel);
+bool event_interrupt_overflow(uint8_t channel);
 
 #endif  // MICROPY_INCLUDED_ATMEL_SAMD_EVENTS_H
diff --git a/ports/atmel-samd/i2s.c b/ports/atmel-samd/i2s.c
new file mode 100644
index 000000000..663ecdd1d
--- /dev/null
+++ b/ports/atmel-samd/i2s.c
@@ -0,0 +1,84 @@
+/*
+ * This file is part of the MicroPython project, http://micropython.org/
+ *
+ * The MIT License (MIT)
+ *
+ * Copyright (c) 2018 Scott Shawcroft for Adafruit Industries
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#include "i2s.h"
+
+#include "clocks.h"
+
+#include "hpl/gclk/hpl_gclk_base.h"
+#ifdef SAMD21
+#include "hpl/pm/hpl_pm_base.h"
+#endif
+
+void turn_on_i2s(void) {
+    // Make sure the I2S peripheral is running so we can see if the resources we need are free.
+    #ifdef SAMD51
+    hri_mclk_set_APBDMASK_I2S_bit(MCLK);
+
+    // Connect the clock units to the 2mhz clock by default. They can't reset without it.
+    connect_gclk_to_peripheral(5, I2S_GCLK_ID_0);
+    connect_gclk_to_peripheral(5, I2S_GCLK_ID_1);
+    #endif
+
+    #ifdef SAMD21
+    _pm_enable_bus_clock(PM_BUS_APBC, I2S);
+    #endif
+}
+
+void i2s_set_enable(bool enable) {
+    while (I2S->SYNCBUSY.bit.ENABLE == 1) {}
+    I2S->CTRLA.bit.ENABLE = enable;
+    while (I2S->SYNCBUSY.bit.ENABLE == 1) {}
+}
+
+void i2s_set_clock_unit_enable(uint8_t clock_unit, bool enable) {
+    while ((I2S->SYNCBUSY.vec.CKEN & (1 << clock_unit)) != 0) {}
+    I2S->CTRLA.vec.CKEN = 1 << clock_unit;
+    while ((I2S->SYNCBUSY.vec.CKEN & (1 << clock_unit)) != 0) {}
+}
+
+void i2s_set_serializer_enable(uint8_t serializer, bool enable) {
+    #ifdef SAMD21
+    while ((I2S->SYNCBUSY.vec.SEREN & (1 << serializer)) != 0) {}
+    if (enable) {
+        I2S->CTRLA.vec.SEREN = 1 << serializer;
+    } else {
+        I2S->CTRLA.vec.SEREN &= ~(1 << serializer);
+    }
+    while ((I2S->SYNCBUSY.vec.SEREN & (1 << serializer)) != 0) {}
+    #endif
+    #ifdef SAMD51
+    if (serializer == 0) {
+        while (I2S->SYNCBUSY.bit.TXEN == 1) {}
+        I2S->CTRLA.bit.TXEN = enable;
+        while (I2S->SYNCBUSY.bit.TXEN == 1) {}
+    } else {
+        while (I2S->SYNCBUSY.bit.RXEN == 1) {}
+        I2S->CTRLA.bit.RXEN = enable;
+        while (I2S->SYNCBUSY.bit.RXEN == 1) {}
+    }
+    #endif
+}
diff --git a/ports/atmel-samd/i2s.h b/ports/atmel-samd/i2s.h
new file mode 100644
index 000000000..7df667b2a
--- /dev/null
+++ b/ports/atmel-samd/i2s.h
@@ -0,0 +1,41 @@
+/*
+ * This file is part of the MicroPython project, http://micropython.org/
+ *
+ * The MIT License (MIT)
+ *
+ * Copyright (c) 2018 Scott Shawcroft for Adafruit Industries
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#ifndef MICROPY_INCLUDED_ATMEL_SAMD_I2S_H
+#define MICROPY_INCLUDED_ATMEL_SAMD_I2S_H
+
+#include <stdbool.h>
+#include <stdint.h>
+
+#include "include/sam.h"
+
+void turn_on_i2s(void);
+
+void i2s_set_enable(bool enable);
+void i2s_set_clock_unit_enable(uint8_t clock, bool enable);
+void i2s_set_serializer_enable(uint8_t serializer, bool enable);
+
+#endif  // MICROPY_INCLUDED_ATMEL_SAMD_I2S_H
diff --git a/shared-bindings/audiobusio/PDMIn.c b/shared-bindings/audiobusio/PDMIn.c
index 6f0a130e8..3e0f3c1fc 100644
--- a/shared-bindings/audiobusio/PDMIn.c
+++ b/shared-bindings/audiobusio/PDMIn.c
@@ -42,18 +42,18 @@
 //|
 //| PDMIn can be used to record an input audio signal on a given set of pins.
 //|
-//| .. class:: PDMIn(clock_pin, data_pin, \*, frequency=16000, bit_depth=8, mono=True, oversample=64, startup_delay=0.11)
+//| .. class:: PDMIn(clock_pin, data_pin, \*, sample_rate=16000, bit_depth=8, mono=True, oversample=64, startup_delay=0.11)
 //|
 //|   Create a PDMIn object associated with the given pins. This allows you to
 //|   record audio signals from the given pins. Individual ports may put further
-//|   restrictions on the recording parameters. The overall frequency is
-//|   determined by `frequency` x ``oversample``, and the total must be 1MHz or
-//|   higher, so `frequency` must be a minimum of 16000.
+//|   restrictions on the recording parameters. The overall sample rate is
+//|   determined by `sample_rate` x ``oversample``, and the total must be 1MHz or
+//|   higher, so `sample_rate` must be a minimum of 16000.
 //|
 //|   :param ~microcontroller.Pin clock_pin: The pin to output the clock to
 //|   :param ~microcontroller.Pin data_pin: The pin to read the data from
-//|   :param int frequency: Target frequency of the resulting samples. Check `frequency` for actual value.
-//|     Minimum frequency is about 16000 Hz.
+//|   :param int sample_rate: Target sample_rate of the resulting samples. Check `sample_rate` for actual value.
+//|     Minimum sample_rate is about 16000 Hz.
 //|   :param int bit_depth: Final number of bits per sample. Must be divisible by 8
 //|   :param bool mono: True when capturing a single channel of audio, captures two channels otherwise
 //|   :param int oversample: Number of single bit samples to decimate into a final sample. Must be divisible by 8
@@ -69,7 +69,7 @@
 //|
 //|     # Prep a buffer to record into
 //|     b = bytearray(200)
-//|     with audiobusio.PDMIn(board.MICROPHONE_CLOCK, board.MICROPHONE_DATA, frequency=16000) as mic:
+//|     with audiobusio.PDMIn(board.MICROPHONE_CLOCK, board.MICROPHONE_DATA, sample_rate=16000) as mic:
 //|         mic.record(b, len(b))
 //|
 //|   Record 16-bit unsigned samples to buffer::
@@ -83,15 +83,15 @@
 //|     b = array.array("H")
 //|     for i in range(200):
 //|         b.append(0)
-//|     with audiobusio.PDMIn(board.MICROPHONE_CLOCK, board.MICROPHONE_DATA, frequency=16000, bit_depth=16) as mic:
+//|     with audiobusio.PDMIn(board.MICROPHONE_CLOCK, board.MICROPHONE_DATA, sample_rate=16000, bit_depth=16) as mic:
 //|         mic.record(b, len(b))
 //|
 STATIC mp_obj_t audiobusio_pdmin_make_new(const mp_obj_type_t *type, size_t n_args, size_t n_kw, const mp_obj_t *pos_args) {
-    enum { ARG_frequency, ARG_bit_depth, ARG_mono, ARG_oversample, ARG_startup_delay };
+    enum { ARG_sample_rate, ARG_bit_depth, ARG_mono, ARG_oversample, ARG_startup_delay };
     mp_map_t kw_args;
     mp_map_init_fixed_table(&kw_args, n_kw, pos_args + n_args);
     static const mp_arg_t allowed_args[] = {
-        { MP_QSTR_frequency,     MP_ARG_KW_ONLY | MP_ARG_INT, {.u_int = 16000} },
+        { MP_QSTR_sample_rate,     MP_ARG_KW_ONLY | MP_ARG_INT, {.u_int = 16000} },
         { MP_QSTR_bit_depth,     MP_ARG_KW_ONLY | MP_ARG_INT, {.u_int = 8} },
         { MP_QSTR_mono,          MP_ARG_KW_ONLY | MP_ARG_BOOL,{.u_bool = true} },
         { MP_QSTR_oversample,    MP_ARG_KW_ONLY | MP_ARG_INT, {.u_int = 64} },
@@ -117,7 +117,7 @@ STATIC mp_obj_t audiobusio_pdmin_make_new(const mp_obj_type_t *type, size_t n_ar
     audiobusio_pdmin_obj_t *self = m_new_obj(audiobusio_pdmin_obj_t);
     self->base.type = &audiobusio_pdmin_type;
 
-    uint32_t frequency = args[ARG_frequency].u_int;
+    uint32_t sample_rate = args[ARG_sample_rate].u_int;
     uint8_t bit_depth = args[ARG_bit_depth].u_int;
     if (bit_depth % 8 != 0) {
         mp_raise_ValueError("Bit depth must be multiple of 8.");
@@ -135,7 +135,7 @@ STATIC mp_obj_t audiobusio_pdmin_make_new(const mp_obj_type_t *type, size_t n_ar
         mp_raise_ValueError("Microphone startup delay must be in range 0.0 to 1.0");
     }
 
-    common_hal_audiobusio_pdmin_construct(self, clock_pin, data_pin, frequency,
+    common_hal_audiobusio_pdmin_construct(self, clock_pin, data_pin, sample_rate,
                                           bit_depth, mono, oversample);
 
     // Wait for the microphone to start up. Some start in 10 msecs; some take as much as 100 msecs.
@@ -215,21 +215,21 @@ STATIC mp_obj_t audiobusio_pdmin_obj_record(mp_obj_t self_obj, mp_obj_t destinat
 }
 MP_DEFINE_CONST_FUN_OBJ_3(audiobusio_pdmin_record_obj, audiobusio_pdmin_obj_record);
 
-//|   .. attribute:: frequency
+//|   .. attribute:: sample_rate
 //|
-//|     The actual frequency of the recording. This may not match the constructed
-//|     frequency due to internal clock limitations.
+//|     The actual sample_rate of the recording. This may not match the constructed
+//|     sample rate due to internal clock limitations.
 //|
-STATIC mp_obj_t audiobusio_pdmin_obj_get_frequency(mp_obj_t self_in) {
+STATIC mp_obj_t audiobusio_pdmin_obj_get_sample_rate(mp_obj_t self_in) {
     audiobusio_pdmin_obj_t *self = MP_OBJ_TO_PTR(self_in);
     raise_error_if_deinited(common_hal_audiobusio_pdmin_deinited(self));
-    return MP_OBJ_NEW_SMALL_INT(common_hal_audiobusio_pdmin_get_frequency(self));
+    return MP_OBJ_NEW_SMALL_INT(common_hal_audiobusio_pdmin_get_sample_rate(self));
 }
-MP_DEFINE_CONST_FUN_OBJ_1(audiobusio_pdmin_get_frequency_obj, audiobusio_pdmin_obj_get_frequency);
+MP_DEFINE_CONST_FUN_OBJ_1(audiobusio_pdmin_get_sample_rate_obj, audiobusio_pdmin_obj_get_sample_rate);
 
-const mp_obj_property_t audiobusio_pdmin_frequency_obj = {
+const mp_obj_property_t audiobusio_pdmin_sample_rate_obj = {
     .base.type = &mp_type_property,
-    .proxy = {(mp_obj_t)&audiobusio_pdmin_get_frequency_obj,
+    .proxy = {(mp_obj_t)&audiobusio_pdmin_get_sample_rate_obj,
               (mp_obj_t)&mp_const_none_obj,
               (mp_obj_t)&mp_const_none_obj},
 };
@@ -240,7 +240,7 @@ STATIC const mp_rom_map_elem_t audiobusio_pdmin_locals_dict_table[] = {
     { MP_ROM_QSTR(MP_QSTR___enter__), MP_ROM_PTR(&default___enter___obj) },
     { MP_ROM_QSTR(MP_QSTR___exit__), MP_ROM_PTR(&audiobusio_pdmin___exit___obj) },
     { MP_ROM_QSTR(MP_QSTR_record), MP_ROM_PTR(&audiobusio_pdmin_record_obj) },
-    { MP_ROM_QSTR(MP_QSTR_frequency), MP_ROM_PTR(&audiobusio_pdmin_frequency_obj) }
+    { MP_ROM_QSTR(MP_QSTR_sample_rate), MP_ROM_PTR(&audiobusio_pdmin_sample_rate_obj) }
 };
 STATIC MP_DEFINE_CONST_DICT(audiobusio_pdmin_locals_dict, audiobusio_pdmin_locals_dict_table);
 
diff --git a/shared-bindings/audiobusio/PDMIn.h b/shared-bindings/audiobusio/PDMIn.h
index a0702c978..eaed59830 100644
--- a/shared-bindings/audiobusio/PDMIn.h
+++ b/shared-bindings/audiobusio/PDMIn.h
@@ -35,13 +35,13 @@ extern const mp_obj_type_t audiobusio_pdmin_type;
 
 void common_hal_audiobusio_pdmin_construct(audiobusio_pdmin_obj_t* self,
     const mcu_pin_obj_t* clock_pin, const mcu_pin_obj_t* data_pin,
-    uint32_t frequency, uint8_t bit_depth, bool mono, uint8_t oversample);
+    uint32_t sample_rate, uint8_t bit_depth, bool mono, uint8_t oversample);
 void common_hal_audiobusio_pdmin_deinit(audiobusio_pdmin_obj_t* self);
 bool common_hal_audiobusio_pdmin_deinited(audiobusio_pdmin_obj_t* self);
 uint32_t common_hal_audiobusio_pdmin_record_to_buffer(audiobusio_pdmin_obj_t* self,
     uint16_t* buffer, uint32_t length);
 uint8_t common_hal_audiobusio_pdmin_get_bit_depth(audiobusio_pdmin_obj_t* self);
-uint32_t common_hal_audiobusio_pdmin_get_frequency(audiobusio_pdmin_obj_t* self);
+uint32_t common_hal_audiobusio_pdmin_get_sample_rate(audiobusio_pdmin_obj_t* self);
 // TODO(tannewt): Add record to file
 
 #endif // MICROPY_INCLUDED_SHARED_BINDINGS_AUDIOBUSIO_AUDIOOUT_H
diff --git a/shared-bindings/audiobusio/__init__.c b/shared-bindings/audiobusio/__init__.c
index 528a8fb31..f7e3a0766 100644
--- a/shared-bindings/audiobusio/__init__.c
+++ b/shared-bindings/audiobusio/__init__.c
@@ -62,7 +62,7 @@
 STATIC const mp_rom_map_elem_t audiobusio_module_globals_table[] = {
     { MP_ROM_QSTR(MP_QSTR___name__), MP_ROM_QSTR(MP_QSTR_audiobusio) },
     { MP_ROM_QSTR(MP_QSTR_I2SOut), MP_ROM_PTR(&audiobusio_i2sout_type) },
-    //{ MP_ROM_QSTR(MP_QSTR_PDMIn), MP_ROM_PTR(&audiobusio_pdmin_type) },
+    { MP_ROM_QSTR(MP_QSTR_PDMIn), MP_ROM_PTR(&audiobusio_pdmin_type) },
 };
 
 STATIC MP_DEFINE_CONST_DICT(audiobusio_module_globals, audiobusio_module_globals_table);