Switch to an MVP-ready 16bit three wire DAC setup

src/audio/audio_task.cpp

@@ -37,15 +37,14 @@ namespace task {
 static const char* kTag = "task";
 static const std::size_t kStackSize = 24 * 1024;
 static const std::size_t kDrainStackSize = 1024;
-static const uint8_t kAudioCore = 0;
 
 auto StartPipeline(Pipeline* pipeline, IAudioSink* sink) -> void {
   // Newly created task will free this.
   AudioTaskArgs* args = new AudioTaskArgs{.pipeline = pipeline, .sink = sink};
 
   ESP_LOGI(kTag, "starting audio pipeline task");
-  xTaskCreatePinnedToCore(&AudioTaskMain, "pipeline", kStackSize, args,
+  xTaskCreate(&AudioTaskMain, "pipeline", kStackSize, args,
-                          kTaskPriorityAudioPipeline, NULL, kAudioCore);
+              kTaskPriorityAudioPipeline, NULL);
 }
 
 auto StartDrain(IAudioSink* sink) -> void {
@@ -151,6 +150,13 @@ void AudioTaskMain(void* args) {
           std::size_t sent = xStreamBufferSend(
               sink->buffer(), sink_stream.data().data(),
               sink_stream.data().size_bytes(), pdMS_TO_TICKS(10));
+          if (sent > 0) {
+            ESP_LOGI(kTag, "sunk %u bytes out of %u (%d %%)", sent,
+                     sink_stream.info().bytes_in_stream,
+                     (int)(((float)sent /
+                            (float)sink_stream.info().bytes_in_stream) *
+                           100));
+          }
           sink_stream.consume(sent);
         }
 

src/audio/include/audio_sink.hpp

@@ -10,7 +10,7 @@ namespace audio {
 class IAudioSink {
  private:
   // TODO: tune. at least about 12KiB seems right for mp3
-  static const std::size_t kDrainBufferSize = 24 * 1024;
+  static const std::size_t kDrainBufferSize = 48 * 1024;
   uint8_t* buffer_;
   StaticStreamBuffer_t* metadata_;
   StreamBufferHandle_t handle_;

src/audio/include/pipeline.hpp

@@ -13,7 +13,7 @@
 
 namespace audio {
 
-static const std::size_t kPipelineBufferSize = 32 * 1024;
+static const std::size_t kPipelineBufferSize = 64 * 1024;
 
 class Pipeline {
  public:

src/codecs/mad.cpp

@@ -1,4 +1,5 @@
 #include "mad.hpp"
+#include <stdint.h>
 
 #include <cstdint>
 
@@ -9,9 +10,9 @@
 
 namespace codecs {
 
-static int scaleTo16Bits(mad_fixed_t sample) {
+static uint32_t scaleToBits(mad_fixed_t sample, uint8_t bits) {
   // Round the bottom bits.
-  sample += (1L << (MAD_F_FRACBITS - 16));
+  sample += (1L << (MAD_F_FRACBITS - bits));
 
   // Clip the leftover bits to within range.
   if (sample >= MAD_F_ONE)
@@ -20,7 +21,7 @@ static int scaleTo16Bits(mad_fixed_t sample) {
     sample = -MAD_F_ONE;
 
   // Quantize.
-  return sample >> (MAD_F_FRACBITS + 1 - 16);
+  return sample >> (MAD_F_FRACBITS + 1 - bits);
 }
 
 MadMp3Decoder::MadMp3Decoder() {
@@ -119,8 +120,19 @@ auto MadMp3Decoder::WriteOutputSamples(cpp::span<std::byte> output)
     }
 
     for (int channel = 0; channel < synth_.pcm.channels; channel++) {
+      // TODO(jacqueline): output 24 bit samples when (if?) we have a downmix
+      // step in the pipeline.
+      /*
+      uint32_t sample_24 =
+          scaleToBits(synth_.pcm.samples[channel][current_sample_], 24);
+      output[output_byte++] = static_cast<std::byte>((sample_24 >> 16) & 0xFF);
+      output[output_byte++] = static_cast<std::byte>((sample_24 >> 8) & 0xFF);
+      output[output_byte++] = static_cast<std::byte>((sample_24)&0xFF);
+      // 24 bit samples must still be aligned to 32 bits. The LSB is ignored.
+      output[output_byte++] = static_cast<std::byte>(0);
+      */
       uint16_t sample_16 =
-          scaleTo16Bits(synth_.pcm.samples[channel][current_sample_]);
+          scaleToBits(synth_.pcm.samples[channel][current_sample_], 16);
       output[output_byte++] = static_cast<std::byte>((sample_16 >> 8) & 0xFF);
       output[output_byte++] = static_cast<std::byte>((sample_16)&0xFF);
     }

src/drivers/dac.cpp

@@ -38,7 +38,7 @@ auto AudioDac::create(GpioExpander* expander)
   channel_config.dma_frame_num = 1024;
   // Triple buffering should be enough to keep samples flowing smoothly.
   // TODO(jacqueline): verify this with 192kHz 32bps.
-  channel_config.dma_desc_num = 8;
+  channel_config.dma_desc_num = 4;
   // channel_config.auto_clear = true;
 
   ESP_ERROR_CHECK(i2s_new_channel(&channel_config, &i2s_handle, NULL));
@@ -53,7 +53,7 @@ auto AudioDac::create(GpioExpander* expander)
   i2s_std_config_t i2s_config = {
       .clk_cfg = dac->clock_config_,
       .slot_cfg = dac->slot_config_,
-      .gpio_cfg = {.mclk = I2S_GPIO_UNUSED,
+      .gpio_cfg = {.mclk = GPIO_NUM_0,
                    .bclk = GPIO_NUM_26,
                    .ws = GPIO_NUM_27,
                    .dout = GPIO_NUM_5,
@@ -167,6 +167,8 @@ bool AudioDac::WaitForPowerState(
 
 auto AudioDac::Reconfigure(BitsPerSample bps, SampleRate rate) -> void {
   if (i2s_active_) {
+    WriteRegister(pcm512x::MUTE, 0b10001);
+    vTaskDelay(1);
     WriteRegister(pcm512x::POWER, 1 << 4);
     i2s_channel_disable(i2s_handle_);
   }
@@ -198,10 +200,10 @@ auto AudioDac::Reconfigure(BitsPerSample bps, SampleRate rate) -> void {
   ESP_ERROR_CHECK(i2s_channel_reconfig_std_clock(i2s_handle_, &clock_config_));
 
   // DAC reconfiguration.
-  //See here : https://e2e.ti.com/support/data_converters/audio_converters/f/64/t/428281
+  // Inspired heavily by https://github.com/tommag/PCM51xx_Arduino (MIT).
-  // for a config example
 
-  // Check that the bit clock (PLL input) is between 1MHz and 50MHz
+  // Check that the bit clock (PLL input) is between 1MHz and 50MHz. It always
+  // should be.
   uint32_t bckFreq = rate * bps * 2;
   if (bckFreq < 1000000 || bckFreq > 50000000) {
     ESP_LOGE(kTag, "bck freq out of range");
@@ -210,43 +212,45 @@ auto AudioDac::Reconfigure(BitsPerSample bps, SampleRate rate) -> void {
 
   // 24 bits is not supported for 44.1kHz and 48kHz.
   if ((rate == SAMPLE_RATE_44_1 || rate == SAMPLE_RATE_48) && bps == BPS_24) {
-    // TODO(jacqueline): implement
+    // TODO(jacqueline): I think this *can* be implemented, but requires a bunch
+    // of maths.
     ESP_LOGE(kTag, "sample rate and bps mismatch");
     return;
   }
 
-  //Initialize system clock from the I2S BCK input
+  // Initialize system clock from the I2S BCK input
-  WriteRegister(pcm512x::ERROR_DETECT, 0x1A);  // Disable clock autoset and ignore SCK detection
+  //  Disable clock autoset and ignore SCK detection
-  WriteRegister(pcm512x::PLL_REF, 0x10);  // Set PLL clock source to BCK
+  WriteRegister(pcm512x::ERROR_DETECT, 0x1A);
-  WriteRegister(pcm512x::DAC_REF, 0x10); // Set DAC clock source to PLL output
+  // Set PLL clock source to BCK
+  WriteRegister(pcm512x::PLL_REF, 0x10);
+  // Set DAC clock source to PLL output
+  WriteRegister(pcm512x::DAC_REF, 0x10);
 
-  //PLL configuration
+  // PLL configuration
   int p, j, d, r;
 
-  //Clock dividers
+  // Clock dividers
   int nmac, ndac, ncp, dosr, idac;
 
-  if (rate == SAMPLE_RATE_11_025 || rate == SAMPLE_RATE_22_05 || rate == SAMPLE_RATE_44_1)
+  if (rate == SAMPLE_RATE_11_025 || rate == SAMPLE_RATE_22_05 ||
-  {
+      rate == SAMPLE_RATE_44_1) {
-    //44.1kHz and derivatives.
+    // 44.1kHz and derivatives.
-    //P = 1, R = 2, D = 0 for all supported combinations.
+    // P = 1, R = 2, D = 0 for all supported combinations.
-    //Set J to have PLL clk = 90.3168 MHz
+    // Set J to have PLL clk = 90.3168 MHz
     p = 1;
     r = 2;
     j = 90316800 / bckFreq / r;
     d = 0;
 
-    //Derive clocks from the 90.3168MHz PLL
+    // Derive clocks from the 90.3168MHz PLL
     nmac = 2;
     ndac = 16;
     ncp = 4;
     dosr = 8;
     idac = 1024;  // DSP clock / sample rate
-  }
+  } else {
-  else
+    // 8kHz and multiples.
-  {
+    // PLL config for a 98.304 MHz PLL clk
-    //8kHz and multiples.
-    //PLL config for a 98.304 MHz PLL clk
     if (bps == BPS_24 && bckFreq > 1536000) {
       p = 3;
     } else if (bckFreq > 12288000) {
@@ -259,11 +263,17 @@ auto AudioDac::Reconfigure(BitsPerSample bps, SampleRate rate) -> void {
     j = 98304000 / (bckFreq / p) / r;
     d = 0;
 
-    //Derive clocks from the 98.304MHz PLL
+    // Derive clocks from the 98.304MHz PLL
     switch (rate) {
-      case SAMPLE_RATE_16: nmac = 6; break;
+      case SAMPLE_RATE_16:
-      case SAMPLE_RATE_32: nmac = 3; break;
+        nmac = 6;
-      default:              nmac = 2; break;
+        break;
+      case SAMPLE_RATE_32:
+        nmac = 3;
+        break;
+      default:
+        nmac = 2;
+        break;
     }
 
     ndac = 16;
@@ -272,7 +282,6 @@ auto AudioDac::Reconfigure(BitsPerSample bps, SampleRate rate) -> void {
     idac = 98304000 / nmac / rate;  // DSP clock / sample rate
   }
 
-
   // Configure PLL
   WriteRegister(pcm512x::PLL_COEFF_0, p - 1);
   WriteRegister(pcm512x::PLL_COEFF_1, j);
@@ -311,6 +320,11 @@ auto AudioDac::Reconfigure(BitsPerSample bps, SampleRate rate) -> void {
   // shut itself down.
   ESP_ERROR_CHECK(i2s_channel_enable(i2s_handle_));
   WriteRegister(pcm512x::POWER, 0);
+
+  if (i2s_active_) {
+    vTaskDelay(1);
+    WriteRegister(pcm512x::MUTE, 0);
+  }
   i2s_active_ = true;
 }