nodes/ndiin/main.cpp

@@ -1,4 +1,3 @@
-#include <algorithm>
 #include <cstring>
 #include <string>
 #include <vector>
@@ -75,7 +74,8 @@ class NDIInNode : public dmf::NodeBase {
                 log("audio mxlCreateFlowWriter failed (status=%d) — continuing without audio", ast);
                 has_audio = false;
             } else {
-                log("audio grain=%u B", audio_cfg.discrete.sliceSizes[0]);
+                log("audio channels=%u  buffer=%u samples",
+                    audio_cfg.continuous.channelCount, audio_cfg.continuous.bufferLength);
             }
         }
 
@@ -104,22 +104,25 @@ class NDIInNode : public dmf::NodeBase {
             if (kind == dmf::NDIReceiver::FrameKind::Video) {
                 have_video = true;
             } else if (kind == dmf::NDIReceiver::FrameKind::Audio && has_audio) {
-                mxlGrainInfo grain{};
+                mxlMutableWrappedMultiBufferSlice slice{};
-                uint8_t* buf = nullptr;
+                if (mxlFlowWriterOpenSamples(audio_writer, audio_index,
-                if (mxlFlowWriterOpenGrain(audio_writer, audio_index, &grain, &buf) == MXL_STATUS_OK) {
+                                             static_cast<size_t>(audio_info.samples), &slice) == MXL_STATUS_OK) {
-                    // Convert float32 planar → int32 interleaved PCM
+                    // MXL audio is float32 planar: each channel occupies its own ring buffer
-                    const int audio_grain_size = audio_cfg.discrete.sliceSizes[0];
+                    // region, accessed at base + c * stride. Fragments handle ring wraparound.
-                    const int capacity = audio_grain_size / (channels * static_cast<int>(sizeof(int32_t)));
+                    const size_t frag0 = slice.base.fragments[0].size / sizeof(float);
-                    const int to_write = std::min(audio_info.samples, capacity);
+                    const size_t frag1 = slice.base.fragments[1].size / sizeof(float);
-                    auto* out = reinterpret_cast<int32_t*>(buf);
+                    for (int c = 0; c < audio_info.channels; ++c) {
-                    for (int s = 0; s < to_write; ++s)
+                        const float* src = &audio_buf[c * audio_info.channel_stride];
-                        for (int c = 0; c < audio_info.channels; ++c) {
+                        auto* dst0 = reinterpret_cast<float*>(
-                            float f = audio_buf[c * audio_info.channel_stride + s];
+                            static_cast<uint8_t*>(slice.base.fragments[0].pointer) + c * slice.stride);
-                            f = std::max(-1.0f, std::min(1.0f, f));
+                        std::memcpy(dst0, src, frag0 * sizeof(float));
-                            *out++ = static_cast<int32_t>(f * 2147483647.0f);
+                        if (frag1 > 0) {
+                            auto* dst1 = reinterpret_cast<float*>(
+                                static_cast<uint8_t*>(slice.base.fragments[1].pointer) + c * slice.stride);
+                            std::memcpy(dst1, src + frag0, frag1 * sizeof(float));
                         }
-                    grain.validSlices = grain.totalSlices;
+                    }
-                    mxlFlowWriterCommitGrain(audio_writer, &grain);
+                    mxlFlowWriterCommitSamples(audio_writer);
                 }
                 audio_index += audio_info.samples;
             }