text: /*
   * This file is part of the MicroPython project, http://micropython.org/
   *
   * The MIT License (MIT)
   *
   * Copyright (c) 2021 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 "audio_dma.h"
  
  #include "shared-bindings/audiocore/RawSample.h"
  #include "shared-bindings/audiocore/WaveFile.h"
  #include "shared-bindings/microcontroller/__init__.h"
  #include "supervisor/background_callback.h"
  
  #include "py/mpstate.h"
  #include "py/runtime.h"
  
  #include "src/rp2_common/hardware_irq/include/hardware/irq.h"
  
  #if CIRCUITPY_AUDIOPWMIO || CIRCUITPY_AUDIOBUSIO
  
  void audio_dma_reset(void) {
      for (size_t channel = 0; channel < NUM_DMA_CHANNELS; channel++) {
          if (MP_STATE_PORT(playing_audio)[channel] == NULL) {
              continue;
          }
  
          audio_dma_stop(MP_STATE_PORT(playing_audio)[channel]);
      }
  }
  
  
  STATIC size_t audio_dma_convert_samples(audio_dma_t *dma, uint8_t *input, uint32_t input_length, uint8_t *output, uint32_t output_length) {
      #pragma GCC diagnostic push
      #pragma GCC diagnostic ignored "-Wcast-align"
  
      uint32_t output_length_used = input_length / dma->sample_spacing;
  
      if (output_length_used > output_length) {
          mp_raise_RuntimeError(translate("Internal audio buffer too small"));
      }
  
      uint32_t out_i = 0;
      if (dma->sample_resolution <= 8 && dma->output_resolution > 8) {
          // reading bytes, writing 16-bit words, so output buffer will be bigger.
  
          output_length_used = output_length * 2;
          if (output_length_used > output_length) {
              mp_raise_RuntimeError(translate("Internal audio buffer too small"));
          }
  
          size_t shift = dma->output_resolution - dma->sample_resolution;
  
          for (uint32_t i = 0; i < input_length; i += dma->sample_spacing) {
              if (dma->signed_to_unsigned) {
                  ((uint16_t *)output)[out_i] = ((uint16_t)((int8_t *)input)[i] + 0x80) << shift;
              } else if (dma->unsigned_to_signed) {
                  ((int16_t *)output)[out_i] = ((int16_t)((uint8_t *)input)[i] - 0x80) << shift;
              } else {
                  ((uint16_t *)output)[out_i] = ((uint16_t)((uint8_t *)input)[i]) << shift;
              }
              out_i += 1;
          }
      } else if (dma->sample_resolution <= 8 && dma->output_resolution <= 8) {
          for (uint32_t i = 0; i < input_length; i += dma->sample_spacing) {
              if (dma->signed_to_unsigned) {
                  ((uint8_t *)output)[out_i] = ((int8_t *)input)[i] + 0x80;
              } else if (dma->unsigned_to_signed) {
                  ((int8_t *)output)[out_i] = ((uint8_t *)input)[i] - 0x80;
              } else {
                  ((uint8_t *)output)[out_i] = ((uint8_t *)input)[i];
              }
              out_i += 1;
          }
      } else if (dma->sample_resolution > 8 && dma->output_resolution > 8) {
          size_t shift = 16 - dma->output_resolution;
          for (uint32_t i = 0; i < input_length / 2; i += dma->sample_spacing) {
              if (dma->signed_to_unsigned) {
                  ((uint16_t *)output)[out_i] = ((int16_t *)input)[i] + 0x8000;
              } else if (dma->unsigned_to_signed) {
                  ((int16_t *)output)[out_i] = ((uint16_t *)input)[i] - 0x8000;
              } else {
                  ((uint16_t *)output)[out_i] = ((uint16_t *)input)[i];
              }
              if (dma->output_resolution < 16) {
                  if (dma->output_signed) {
                      ((int16_t *)output)[out_i] = ((int16_t *)output)[out_i] >> shift;
                  } else {
                      ((uint16_t *)output)[out_i] = ((uint16_t *)output)[out_i] >> shift;
                  }
              }
              out_i += 1;
          }
      } else {
          // (dma->sample_resolution > 8 && dma->output_resolution <= 8)
          // Not currently used, but might be in the future.
          mp_raise_RuntimeError(translate("Audio conversion not implemented"));
      }
      #pragma GCC diagnostic pop
      return output_length_used;
  }
  
  // buffer_idx is 0 or 1.
  STATIC void audio_dma_load_next_block(audio_dma_t *dma, size_t buffer_idx) {
      size_t dma_channel = dma->channel[buffer_idx];
  
      audioio_get_buffer_result_t get_buffer_result;
      uint8_t *sample_buffer;
      uint32_t sample_buffer_length;
      get_buffer_result = audiosample_get_buffer(dma->sample,
          dma->single_channel_output, dma->audio_channel, &sample_buffer, &sample_buffer_length);
  
      if (get_buffer_result == GET_BUFFER_ERROR) {
          audio_dma_stop(dma);
          return;
      }
  
      // Convert the sample format resolution and signedness, as necessary.
      // The input sample buffer is what was read from a file, Mixer, or a raw sample buffer.
      // The output buffer is one of the DMA buffers (passed in).
  
      size_t output_length_used = audio_dma_convert_samples(
          dma, sample_buffer, sample_buffer_length,
          dma->buffer[buffer_idx], dma->buffer_length[buffer_idx]);
  
      dma_channel_set_read_addr(dma_channel, dma->buffer[buffer_idx], false /* trigger */);
      dma_channel_set_trans_count(dma_channel, output_length_used / dma->output_size, false /* trigger */);
  
      if (get_buffer_result == GET_BUFFER_DONE) {
          if (dma->loop) {
              audiosample_reset_buffer(dma->sample, dma->single_channel_output, dma->audio_channel);
          } else {
              // Set channel trigger to ourselves so we don't keep going.
              dma_channel_hw_t *c = &dma_hw->ch[dma_channel];
              c->al1_ctrl =
                  (c->al1_ctrl & ~DMA_CH0_CTRL_TRIG_CHAIN_TO_BITS) |
                  (dma_channel << DMA_CH0_CTRL_TRIG_CHAIN_TO_LSB);
  
              if (output_length_used == 0 &&
                  !dma_channel_is_busy(dma->channel[0]) &&
                  !dma_channel_is_busy(dma->channel[1])) {
                  // No data has been read, and both DMA channels have now finished, so it's safe to stop.
                  audio_dma_stop(dma);
                  dma->playing_in_progress = false;
              }
          }
      }
  }
  
  // Playback should be shutdown before calling this.
  audio_dma_result audio_dma_setup_playback(
      audio_dma_t *dma,
      mp_obj_t sample,
      bool loop,
      bool single_channel_output,
      uint8_t audio_channel,
      bool output_signed,
      uint8_t output_resolution,
      uint32_t output_register_address,
      uint8_t dma_trigger_source) {
  
      // Use two DMA channels to play because the DMA can't wrap to itself without the
      // buffer being power of two aligned.
      int dma_channel_0_maybe = dma_claim_unused_channel(false);
      if (dma_channel_0_maybe < 0) {
          return AUDIO_DMA_DMA_BUSY;
      }
  
      int dma_channel_1_maybe = dma_claim_unused_channel(false);
      if (dma_channel_1_maybe < 0) {
          dma_channel_unclaim((uint)dma_channel_0_maybe);
          return AUDIO_DMA_DMA_BUSY;
      }
  
      dma->channel[0] = (uint8_t)dma_channel_0_maybe;
      dma->channel[1] = (uint8_t)dma_channel_1_maybe;
  
      dma->sample = sample;
      dma->loop = loop;
      dma->single_channel_output = single_channel_output;
      dma->audio_channel = audio_channel;
      dma->signed_to_unsigned = false;
      dma->unsigned_to_signed = false;
      dma->output_signed = output_signed;
      dma->sample_spacing = 1;
      dma->output_resolution = output_resolution;
      dma->sample_resolution = audiosample_bits_per_sample(sample);
      dma->output_register_address = output_register_address;
  
      audiosample_reset_buffer(sample, single_channel_output, audio_channel);
  
  
      bool single_buffer;  // True if data fits in one single buffer.
  
      bool samples_signed;
      uint32_t max_buffer_length;
      audiosample_get_buffer_structure(sample, single_channel_output, &single_buffer, &samples_signed,
          &max_buffer_length, &dma->sample_spacing);
  
      // Check to see if we have to scale the resolution up.
      if (dma->sample_resolution <= 8 && dma->output_resolution > 8) {
          max_buffer_length *= 2;
      }
      if (output_signed != samples_signed ||
          dma->sample_spacing > 1 ||
          (dma->sample_resolution != dma->output_resolution)) {
          max_buffer_length /= dma->sample_spacing;
      }
  
      dma->buffer[0] = (uint8_t *)m_realloc(dma->buffer[0], max_buffer_length);
      dma->buffer_length[0] = max_buffer_length;
      if (dma->buffer[0] == NULL) {
          return AUDIO_DMA_MEMORY_ERROR;
      }
  
      if (!single_buffer) {
          dma->buffer[1] = (uint8_t *)m_realloc(dma->buffer[1], max_buffer_length);
          dma->buffer_length[1] = max_buffer_length;
          if (dma->buffer[1] == NULL) {
              return AUDIO_DMA_MEMORY_ERROR;
          }
      }
  
      dma->signed_to_unsigned = !output_signed && samples_signed;
      dma->unsigned_to_signed = output_signed && !samples_signed;
  
      if (output_resolution > 8) {
          dma->output_size = 2;
      } else {
          dma->output_size = 1;
      }
      // Transfer both channels at once.
      if (!single_channel_output && audiosample_channel_count(sample) == 2) {
          dma->output_size *= 2;
      }
      enum dma_channel_transfer_size dma_size = DMA_SIZE_8;
      if (dma->output_size == 2) {
          dma_size = DMA_SIZE_16;
      } else if (dma->output_size == 4) {
          dma_size = DMA_SIZE_32;
      }
  
      for (size_t i = 0; i < 2; i++) {
          dma_channel_config c = dma_channel_get_default_config(dma->channel[i]);
          channel_config_set_transfer_data_size(&c, dma_size);
          channel_config_set_dreq(&c, dma_trigger_source);
          channel_config_set_read_increment(&c, true);
          channel_config_set_write_increment(&c, false);
  
          // Chain to the other channel by default.
          channel_config_set_chain_to(&c, dma->channel[(i + 1) % 2]);
          dma_channel_set_config(dma->channel[i], &c, false /* trigger */);
  
          dma_channel_set_write_addr(dma->channel[i], (void *)output_register_address, false /* trigger */);
      }
  
      // We keep the audio_dma_t for internal use and the sample as a root pointer because it
      // contains the audiodma structure.
      MP_STATE_PORT(playing_audio)[dma->channel[0]] = dma;
      MP_STATE_PORT(playing_audio)[dma->channel[1]] = dma;
  
      // Load the first two blocks up front.
      audio_dma_load_next_block(dma, 0);
      if (!single_buffer) {
          audio_dma_load_next_block(dma, 1);
      }
  
      // Special case the DMA for a single buffer. It's commonly used for a single wave length of sound
      // and may be short. Therefore, we use DMA chaining to loop quickly without involving interrupts.
      // On the RP2040 we chain by having a second DMA writing to the config registers of the first.
      // Read and write addresses change with DMA so we need to reset the read address back to the
      // start of the sample.
      if (single_buffer) {
          dma_channel_config c = dma_channel_get_default_config(dma->channel[1]);
          channel_config_set_transfer_data_size(&c, DMA_SIZE_32);
          channel_config_set_dreq(&c, 0x3f); // dma as fast as possible
          channel_config_set_read_increment(&c, false);
          channel_config_set_write_increment(&c, false);
          channel_config_set_chain_to(&c, dma->channel[1]); // Chain to ourselves so we stop.
          dma_channel_configure(dma->channel[1], &c,
              &dma_hw->ch[dma->channel[0]].al3_read_addr_trig, // write address
              &dma->buffer[0], // read address
              1, // transaction count
              false); // trigger
      } else {
          // Enable our DMA channels on DMA_IRQ_0 to the CPU. This will wake us up when
          // we're WFI.
          dma_hw->inte0 |= (1 << dma->channel[0]) | (1 << dma->channel[1]);
          irq_set_mask_enabled(1 << DMA_IRQ_0, true);
      }
  
      dma->playing_in_progress = true;
      dma_channel_start(dma->channel[0]);
  
      return AUDIO_DMA_OK;
  }
  
  void audio_dma_stop(audio_dma_t *dma) {
      // Disable our interrupts.
      uint32_t channel_mask = 0;
      if (dma->channel[0] < NUM_DMA_CHANNELS) {
          channel_mask |= 1 << dma->channel[0];
      }
      if (dma->channel[1] < NUM_DMA_CHANNELS) {
          channel_mask |= 1 << dma->channel[1];
      }
      dma_hw->inte0 &= ~channel_mask;
      irq_set_mask_enabled(1 << DMA_IRQ_0, false);
  
      // Run any remaining audio tasks because we remove ourselves from
      // playing_audio.
      RUN_BACKGROUND_TASKS;
  
      for (size_t i = 0; i < 2; i++) {
          size_t channel = dma->channel[i];
          if (channel == NUM_DMA_CHANNELS) {
              // Channel not in use.
              continue;
          }
  
          dma_channel_config c = dma_channel_get_default_config(dma->channel[i]);
          channel_config_set_enable(&c, false);
          dma_channel_set_config(channel, &c, false /* trigger */);
  
          if (dma_channel_is_busy(channel)) {
              dma_channel_abort(channel);
          }
  
          dma_channel_set_read_addr(channel, NULL, false /* trigger */);
          dma_channel_set_write_addr(channel, NULL, false /* trigger */);
          dma_channel_set_trans_count(channel, 0, false /* trigger */);
          dma_channel_unclaim(channel);
          MP_STATE_PORT(playing_audio)[channel] = NULL;
          dma->channel[i] = NUM_DMA_CHANNELS;
      }
      dma->playing_in_progress = false;
  
      // Hold onto our buffers.
  }
  
  // To pause we simply stop the DMA. It is the responsibility of the output peripheral
  // to hold the previous value.
  void audio_dma_pause(audio_dma_t *dma) {
      dma_hw->ch[dma->channel[0]].al1_ctrl &= ~DMA_CH0_CTRL_TRIG_EN_BITS;
      dma_hw->ch[dma->channel[1]].al1_ctrl &= ~DMA_CH1_CTRL_TRIG_EN_BITS;
  }
  
  void audio_dma_resume(audio_dma_t *dma) {
      // Always re-enable the non-busy channel first so it's ready to continue when the busy channel
      // finishes and chains to it. (An interrupt could make the time between enables long.)
      if (dma_channel_is_busy(dma->channel[0])) {
          dma_hw->ch[dma->channel[1]].al1_ctrl |= DMA_CH1_CTRL_TRIG_EN_BITS;
          dma_hw->ch[dma->channel[0]].al1_ctrl |= DMA_CH0_CTRL_TRIG_EN_BITS;
      } else {
          dma_hw->ch[dma->channel[0]].al1_ctrl |= DMA_CH0_CTRL_TRIG_EN_BITS;
          dma_hw->ch[dma->channel[1]].al1_ctrl |= DMA_CH1_CTRL_TRIG_EN_BITS;
      }
  }
  
  bool audio_dma_get_paused(audio_dma_t *dma) {
      if (dma->channel[0] >= NUM_DMA_CHANNELS) {
          return false;
      }
      uint32_t control = dma_hw->ch[dma->channel[0]].ctrl_trig;
  
      return (control & DMA_CH0_CTRL_TRIG_EN_BITS) == 0;
  }
  
  void audio_dma_init(audio_dma_t *dma) {
      dma->buffer[0] = NULL;
      dma->buffer[1] = NULL;
  
      dma->channel[0] = NUM_DMA_CHANNELS;
      dma->channel[1] = NUM_DMA_CHANNELS;
  }
  
  void audio_dma_deinit(audio_dma_t *dma) {
      m_free(dma->buffer[0]);
      dma->buffer[0] = NULL;
  
      m_free(dma->buffer[1]);
      dma->buffer[1] = NULL;
  }
  
  bool audio_dma_get_playing(audio_dma_t *dma) {
      if (dma->channel[0] == NUM_DMA_CHANNELS) {
          return false;
      }
      return dma->playing_in_progress;
  }
  
  // WARN(tannewt): DO NOT print from here, or anything it calls. Printing calls
  // background tasks such as this and causes a stack overflow.
  // NOTE(dhalbert): I successfully printed from here while debugging.
  // So it's possible, but be careful.
  STATIC void dma_callback_fun(void *arg) {
      audio_dma_t *dma = arg;
      if (dma == NULL) {
          return;
      }
  
      common_hal_mcu_disable_interrupts();
      uint32_t channels_to_load_mask = dma->channels_to_load_mask;
      dma->channels_to_load_mask = 0;
      common_hal_mcu_enable_interrupts();
  
      // Load the blocks for the requested channels.
      uint32_t channel = 0;
      while (channels_to_load_mask) {
          if (channels_to_load_mask & 1) {
              if (dma->channel[0] == channel) {
                  audio_dma_load_next_block(dma, 0);
              }
              if (dma->channel[1] == channel) {
                  audio_dma_load_next_block(dma, 1);
              }
          }
          channels_to_load_mask >>= 1;
          channel++;
      }
  }
  
  void isr_dma_0(void) {
      for (size_t i = 0; i < NUM_DMA_CHANNELS; i++) {
          uint32_t mask = 1 << i;
          if ((dma_hw->intr & mask) != 0 && MP_STATE_PORT(playing_audio)[i] != NULL) {
              audio_dma_t *dma = MP_STATE_PORT(playing_audio)[i];
              // Record all channels whose DMA has completed; they need loading.
              dma->channels_to_load_mask |= mask;
              background_callback_add(&dma->callback, dma_callback_fun, (void *)dma);
              dma_hw->ints0 = mask;
          }
      }
  }
  
  #endif
  