Fix resampler issue, do a little performance tuning

4 files changed, 34 insertions, 22 deletions

diff --git a/src/audio/fatfs_audio_input.cpp b/src/audio/fatfs_audio_input.cpp
index 0c3ef20d..73586f09 100644
--- a/src/audio/fatfs_audio_input.cpp
+++ b/src/audio/fatfs_audio_input.cpp
@@ -30,6 +30,7 @@
 #include "freertos/portmacro.h"
 #include "freertos/projdefs.h"
 #include "future_fetcher.hpp"
+#include "idf_additions.h"
 #include "span.hpp"
 #include "stream_info.hpp"
 #include "tag_parser.hpp"
@@ -40,8 +41,8 @@ static const char* kTag = "SRC";
 
 namespace audio {
 
-static constexpr UINT kFileBufferSize = 4096 * 2;
-static constexpr UINT kStreamerBufferSize = 4096;
+static constexpr UINT kFileBufferSize = 8 * 1024;
+static constexpr UINT kStreamerBufferSize = 64 * 1024;
 
 static StreamBufferHandle_t sForwardDest = nullptr;
 
@@ -143,7 +144,9 @@ FatfsAudioInput::FatfsAudioInput(
     : IAudioSource(),
       tag_parser_(tag_parser),
       has_data_(xSemaphoreCreateBinary()),
-      streamer_buffer_(xStreamBufferCreate(kStreamerBufferSize, 1)),
+      streamer_buffer_(xStreamBufferCreateWithCaps(kStreamerBufferSize,
+                                                   1,
+                                                   MALLOC_CAP_SPIRAM)),
       streamer_(new FileStreamer(streamer_buffer_, has_data_)),
       input_buffer_(new RawStream(kFileBufferSize)),
       source_mutex_(),
diff --git a/src/audio/i2s_audio_output.cpp b/src/audio/i2s_audio_output.cpp
index e8aa8975..e53dbe2a 100644
--- a/src/audio/i2s_audio_output.cpp
+++ b/src/audio/i2s_audio_output.cpp
@@ -117,18 +117,10 @@ auto I2SAudioOutput::AdjustVolumeDown() -> bool {
 
 auto I2SAudioOutput::PrepareFormat(const StreamInfo::Pcm& orig)
     -> StreamInfo::Pcm {
-  /*
-return StreamInfo::Pcm{
-    .channels = std::min<uint8_t>(orig.channels, 2),
-    .bits_per_sample = std::clamp<uint8_t>(orig.bits_per_sample, 16, 32),
-    .sample_rate = std::clamp<uint32_t>(orig.sample_rate, 8000, 96000),
-};
-  */
   return StreamInfo::Pcm{
       .channels = std::min<uint8_t>(orig.channels, 2),
-      .bits_per_sample = 16,
-      //.sample_rate = std::clamp<uint32_t>(orig.sample_rate, 8000, 96000),
-      .sample_rate = 44100,
+      .bits_per_sample = std::clamp<uint8_t>(orig.bits_per_sample, 16, 32),
+      .sample_rate = std::clamp<uint32_t>(orig.sample_rate, 8000, 96000),
   };
 }
 
diff --git a/src/audio/resample.cpp b/src/audio/resample.cpp
index aa4c8f2a..7accd0a1 100644
--- a/src/audio/resample.cpp
+++ b/src/audio/resample.cpp
@@ -17,8 +17,8 @@ namespace audio {
 static constexpr char kTag[] = "resample";
 
 static constexpr double kLowPassRatio = 0.5;
-static constexpr size_t kNumFilters = 8;
-static constexpr size_t kTapsPerFilter = 8;
+static constexpr size_t kNumFilters = 64;
+static constexpr size_t kTapsPerFilter = 16;
 
 typedef std::array<float, kTapsPerFilter> Filter;
 static std::array<Filter, kNumFilters + 1> sFilters{};
@@ -64,12 +64,15 @@ auto Resampler::Process(cpp::span<const sample::Sample> input,
   size_t input_frames = input.size() / num_channels_;
   size_t output_frames = output.size() / num_channels_;
 
-  int half_taps = kTapsPerFilter / 2, i;
+  int half_taps = kTapsPerFilter / 2;
   while (output_frames > 0) {
     if (output_offset_ >= input_index_ - half_taps) {
       if (input_frames > 0) {
+        // Check whether the channel buffers will overflow with the addition of
+        // this sample. If so, we need to move the remaining contents back to
+        // the beginning of the buffer.
         if (input_index_ == channel_buffer_size_) {
-          for (i = 0; i < num_channels_; ++i) {
+          for (int i = 0; i < num_channels_; ++i) {
             memmove(channel_buffers_[i],
                     channel_buffers_[i] + channel_buffer_size_ - kTapsPerFilter,
                     kTapsPerFilter * sizeof(float));
@@ -79,21 +82,23 @@ auto Resampler::Process(cpp::span<const sample::Sample> input,
           input_index_ -= channel_buffer_size_ - kTapsPerFilter;
         }
 
-        for (i = 0; i < num_channels_; ++i) {
+        for (int i = 0; i < num_channels_; ++i) {
           channel_buffers_[i][input_index_] =
               sample::ToFloat(input[samples_used++]);
         }
 
         input_index_++;
         input_frames--;
-      } else
+      } else {
         break;
+      }
     } else {
-      for (i = 0; i < num_channels_; i++) {
+      for (int i = 0; i < num_channels_; i++) {
         output[samples_produced++] = sample::FromFloat(Subsample(i));
       }
 
       output_offset_ += (1.0f / factor_);
+      output_frames--;
     }
   }
 
@@ -160,8 +165,20 @@ auto Resampler::Subsample(int channel) -> float {
   source = source.subspan(offset_integral);
   float offset_fractional = output_offset_ - offset_integral;
 
-  int filter_index = offset_fractional * kNumFilters;
+  /*
+// no interpolate
+size_t filter_index = std::floor(offset_fractional * kNumFilters + 0.5f);
+//ESP_LOGI(kTag, "selected filter %u of %u", filter_index, kNumFilters);
+int start_offset = kTapsPerFilter / 2 + 1;
+//ESP_LOGI(kTag, "using offset of %i, length %u", start_offset, kTapsPerFilter);
+
+return ApplyFilter(
+    sFilters[filter_index],
+    {source.data() - start_offset, kTapsPerFilter});
+  */
+
   offset_fractional *= kNumFilters;
+  int filter_index = std::floor(offset_fractional);
 
   sum1 = ApplyFilter(sFilters[filter_index],
                      {source.data() - kTapsPerFilter / 2 + 1, kTapsPerFilter});
diff --git a/src/audio/sink_mixer.cpp b/src/audio/sink_mixer.cpp
index a2bf229b..5a5a8616 100644
--- a/src/audio/sink_mixer.cpp
+++ b/src/audio/sink_mixer.cpp
@@ -195,7 +195,7 @@ auto SinkMixer::Resample(InputStream& in, OutputStream& out) -> bool {
                                  out.data_as<sample::Sample>(), false);
 
   in.consume(res.first * sizeof(sample::Sample));
-  out.add(res.first * sizeof(sample::Sample));
+  out.add(res.second * sizeof(sample::Sample));
 
   return res.first == 0 && res.second == 0;
 }