Audio experiments

Initial audio support
2026-05-05 12:18:42 +10:00 · 2026-05-04 14:32:29 +10:00
18 changed files with 1627 additions and 21 deletions
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -18,6 +18,8 @@ if(NOT EXISTS "${GPUDIRECT_DIR}/lib/x64/dvp.lib")
 endif()

 set(APP_SOURCES
+	"${APP_DIR}/AudioSupport.cpp"
+	"${APP_DIR}/AudioSupport.h"
 	"${APP_DIR}/ControlServer.cpp"
 	"${APP_DIR}/ControlServer.h"
 	"${APP_DIR}/DeckLinkAPI_i.c"
@@ -147,6 +149,21 @@ endif()

 add_test(NAME OscServerTests COMMAND OscServerTests)

+add_executable(AudioSupportTests
+	"${APP_DIR}/AudioSupport.cpp"
+	"${CMAKE_CURRENT_SOURCE_DIR}/tests/AudioSupportTests.cpp"
+)
+
+target_include_directories(AudioSupportTests PRIVATE
+	"${APP_DIR}"
+)
+
+if(MSVC)
+	target_compile_options(AudioSupportTests PRIVATE /W3)
+endif()
+
+add_test(NAME AudioSupportTests COMMAND AudioSupportTests)
+
 add_custom_command(TARGET LoopThroughWithOpenGLCompositing POST_BUILD
 	COMMAND ${CMAKE_COMMAND} -E copy_if_different
 		"${GPUDIRECT_DIR}/bin/x64/dvp.dll"
--- a/README.md
+++ b/README.md
@@ -126,12 +126,18 @@ Current native test coverage includes:
  "outputFrameRate": "59.94",
  "autoReload": true,
  "maxTemporalHistoryFrames": 12,
+  "audioEnabled": true,
+  "audioChannelCount": 2,
+  "audioSampleRate": 48000,
+  "audioDelayMode": "matchVideoPreroll",
  "enableExternalKeying": true
 }
 ```

 `inputVideoFormat`/`inputFrameRate` select the DeckLink capture mode. `outputVideoFormat`/`outputFrameRate` select the playout mode. The shader stack runs at input resolution and the final rendered frame is scaled once into the configured output mode. Common examples include `720p`/`50`, `720p`/`59.94`, `1080i`/`50`, `1080i`/`59.94`, `1080p`/`25`, `1080p`/`50`, `1080p`/`59.94`, and `2160p`/`59.94`, depending on card support.

+`audioEnabled` enables embedded stereo 48 kHz PCM pass-through. Audio is delayed to match the scheduled video preroll and the synchronized level/spectrum data is exposed to shaders.
+
 Legacy `videoFormat` and `frameRate` keys are still accepted and apply to both input and output unless the explicit input/output keys are present.

 The control UI is available at:
--- a/SHADER_CONTRACT.md
+++ b/SHADER_CONTRACT.md
@@ -125,6 +125,11 @@ struct ShaderContext
    float bypass;
    int sourceHistoryLength;
    int temporalHistoryLength;
+    float2 audioRms;
+    float2 audioPeak;
+    float audioMonoRms;
+    float audioMonoPeak;
+    float4 audioBands;
 };
 ```

@@ -140,6 +145,11 @@ Fields:
 - `bypass`: `1.0` when the layer is bypassed, otherwise `0.0`.
 - `sourceHistoryLength`: number of usable source-history frames currently available.
 - `temporalHistoryLength`: number of usable temporal frames currently available for this layer.
+- `audioRms`: left/right RMS level for the audio block synchronized with the rendered output frame.
+- `audioPeak`: left/right peak level for the same synchronized audio block.
+- `audioMonoRms`: mono RMS level derived from left/right.
+- `audioMonoPeak`: mono peak level derived from left/right.
+- `audioBands`: four smoothed, normalized low-to-high frequency bands.

 ## Helper Functions

@@ -149,6 +159,8 @@ The wrapper provides:
 float4 sampleVideo(float2 uv);
 float4 sampleSourceHistory(int framesAgo, float2 uv);
 float4 sampleTemporalHistory(int framesAgo, float2 uv);
+float4 sampleAudioWaveform(float x);
+float4 sampleAudioSpectrum(float x);
 ```

 `sampleVideo` samples the live decoded source video.
@@ -157,6 +169,10 @@ float4 sampleTemporalHistory(int framesAgo, float2 uv);

 `sampleTemporalHistory` samples previous pre-layer input frames for temporal shaders that request `preLayerInput` history. `framesAgo` is clamped into the available range. If no temporal history is available, it falls back to `sampleVideo`.

+`sampleAudioWaveform` samples the current synchronized audio waveform texture. `x` is normalized `0..1`; returned waveform channels are encoded from `-1..1` into `0..1`.
+
+`sampleAudioSpectrum` samples the current synchronized audio spectrum texture. Values are normalized `0..1`.
+
 Example:

 ```slang
--- a/apps/LoopThroughWithOpenGLCompositing/AudioSupport.cpp
+++ b/apps/LoopThroughWithOpenGLCompositing/AudioSupport.cpp
@@ -0,0 +1,206 @@
+#include "AudioSupport.h"
+
+#include <algorithm>
+#include <cmath>
+#include <iterator>
+#include <limits>
+
+namespace
+{
+constexpr float kInt32ToFloat = 1.0f / 2147483648.0f;
+constexpr std::size_t kAnalysisWindowSamples = 1024;
+constexpr std::size_t kMaxBufferedAudioFrames = kAudioSampleRate * 10;
+
+float Clamp01(float value)
+{
+	return std::max(0.0f, std::min(1.0f, value));
+}
+
+float SampleToFloat(int32_t sample)
+{
+	return std::max(-1.0f, std::min(1.0f, static_cast<float>(sample) * kInt32ToFloat));
+}
+
+float GoertzelMagnitude(const std::vector<float>& samples, float frequency)
+{
+	if (samples.empty())
+		return 0.0f;
+
+	const double omega = 2.0 * 3.14159265358979323846 * static_cast<double>(frequency) / static_cast<double>(kAudioSampleRate);
+	const double coefficient = 2.0 * std::cos(omega);
+	double q0 = 0.0;
+	double q1 = 0.0;
+	double q2 = 0.0;
+
+	for (float sample : samples)
+	{
+		q0 = coefficient * q1 - q2 + static_cast<double>(sample);
+		q2 = q1;
+		q1 = q0;
+	}
+
+	const double power = q1 * q1 + q2 * q2 - coefficient * q1 * q2;
+	return static_cast<float>(std::sqrt(std::max(0.0, power)) / static_cast<double>(samples.size()));
+}
+}
+
+uint64_t AudioSampleTimeForVideoFrame(uint64_t videoFrameIndex, uint64_t frameDuration, uint64_t frameTimescale, uint64_t audioSampleRate)
+{
+	if (frameTimescale == 0)
+		return 0;
+
+	const uint64_t numerator = videoFrameIndex * frameDuration * audioSampleRate;
+	return (numerator + frameTimescale / 2) / frameTimescale;
+}
+
+unsigned AudioSamplesForVideoFrame(uint64_t videoFrameIndex, uint64_t frameDuration, uint64_t frameTimescale, uint64_t audioSampleRate)
+{
+	const uint64_t start = AudioSampleTimeForVideoFrame(videoFrameIndex, frameDuration, frameTimescale, audioSampleRate);
+	const uint64_t end = AudioSampleTimeForVideoFrame(videoFrameIndex + 1, frameDuration, frameTimescale, audioSampleRate);
+	return static_cast<unsigned>(end > start ? end - start : 0);
+}
+
+void AudioDelayBuffer::Reset(unsigned delaySampleFrames)
+{
+	std::lock_guard<std::mutex> lock(mMutex);
+	mSamples.clear();
+	mSamples.resize(static_cast<std::size_t>(delaySampleFrames) * kAudioChannelCount, 0);
+	mUnderrunCount = 0;
+}
+
+void AudioDelayBuffer::PushInterleaved(const int32_t* samples, std::size_t sampleFrameCount)
+{
+	if (!samples || sampleFrameCount == 0)
+		return;
+
+	std::lock_guard<std::mutex> lock(mMutex);
+	const std::size_t sampleCount = sampleFrameCount * kAudioChannelCount;
+	for (std::size_t index = 0; index < sampleCount; ++index)
+		mSamples.push_back(samples[index]);
+
+	const std::size_t maxSamples = kMaxBufferedAudioFrames * kAudioChannelCount;
+	while (mSamples.size() > maxSamples)
+		mSamples.pop_front();
+}
+
+AudioFrameBlock AudioDelayBuffer::Pop(std::size_t sampleFrameCount, bool& underrun)
+{
+	AudioFrameBlock block;
+	block.interleavedSamples.resize(sampleFrameCount * kAudioChannelCount, 0);
+
+	std::lock_guard<std::mutex> lock(mMutex);
+	const std::size_t requestedSamples = sampleFrameCount * kAudioChannelCount;
+	underrun = mSamples.size() < requestedSamples;
+	if (underrun)
+		++mUnderrunCount;
+
+	const std::size_t availableSamples = std::min(requestedSamples, mSamples.size());
+	for (std::size_t index = 0; index < availableSamples; ++index)
+	{
+		block.interleavedSamples[index] = mSamples.front();
+		mSamples.pop_front();
+	}
+
+	return block;
+}
+
+unsigned AudioDelayBuffer::BufferedSampleFrames() const
+{
+	std::lock_guard<std::mutex> lock(mMutex);
+	return static_cast<unsigned>(mSamples.size() / kAudioChannelCount);
+}
+
+uint64_t AudioDelayBuffer::UnderrunCount() const
+{
+	std::lock_guard<std::mutex> lock(mMutex);
+	return mUnderrunCount;
+}
+
+void AudioAnalyzer::Reset()
+{
+	mMonoHistory.clear();
+	mSmoothedBands = { 0.0f, 0.0f, 0.0f, 0.0f };
+	mCurrent = AudioAnalysisSnapshot();
+}
+
+AudioAnalysisSnapshot AudioAnalyzer::Analyze(const AudioFrameBlock& block)
+{
+	AudioAnalysisSnapshot next;
+	double sumSquares[2] = { 0.0, 0.0 };
+	float peak[2] = { 0.0f, 0.0f };
+	double monoSumSquares = 0.0;
+	float monoPeak = 0.0f;
+	const std::size_t frames = block.frameCount();
+
+	for (std::size_t frame = 0; frame < frames; ++frame)
+	{
+		const float left = SampleToFloat(block.interleavedSamples[frame * 2]);
+		const float right = SampleToFloat(block.interleavedSamples[frame * 2 + 1]);
+		const float mono = (left + right) * 0.5f;
+
+		sumSquares[0] += static_cast<double>(left) * left;
+		sumSquares[1] += static_cast<double>(right) * right;
+		peak[0] = std::max(peak[0], std::abs(left));
+		peak[1] = std::max(peak[1], std::abs(right));
+		monoSumSquares += static_cast<double>(mono) * mono;
+		monoPeak = std::max(monoPeak, std::abs(mono));
+
+		mMonoHistory.push_back(mono);
+		while (mMonoHistory.size() > kAnalysisWindowSamples)
+			mMonoHistory.pop_front();
+	}
+
+	if (frames > 0)
+	{
+		next.rms[0] = static_cast<float>(std::sqrt(sumSquares[0] / static_cast<double>(frames)));
+		next.rms[1] = static_cast<float>(std::sqrt(sumSquares[1] / static_cast<double>(frames)));
+		next.peak[0] = peak[0];
+		next.peak[1] = peak[1];
+		next.monoRms = static_cast<float>(std::sqrt(monoSumSquares / static_cast<double>(frames)));
+		next.monoPeak = monoPeak;
+	}
+
+	std::vector<float> window(mMonoHistory.begin(), mMonoHistory.end());
+	const float bandFrequencies[4] = { 90.0f, 300.0f, 1200.0f, 5000.0f };
+	for (std::size_t band = 0; band < next.bands.size(); ++band)
+	{
+		const float raw = Clamp01(GoertzelMagnitude(window, bandFrequencies[band]) * 8.0f);
+		const float smoothing = raw > mSmoothedBands[band] ? 0.45f : 0.12f;
+		mSmoothedBands[band] = mSmoothedBands[band] + (raw - mSmoothedBands[band]) * smoothing;
+		next.bands[band] = Clamp01(mSmoothedBands[band]);
+	}
+
+	for (unsigned x = 0; x < kAudioTextureWidth; ++x)
+	{
+		float mono = 0.0f;
+		if (!mMonoHistory.empty())
+		{
+			const std::size_t historyIndex = static_cast<std::size_t>(
+				(static_cast<uint64_t>(x) * static_cast<uint64_t>(mMonoHistory.size())) / kAudioTextureWidth);
+			auto it = mMonoHistory.begin();
+			std::advance(it, std::min(historyIndex, mMonoHistory.size() - 1));
+			mono = *it;
+		}
+
+		const std::size_t waveformOffset = x * 4;
+		next.texture[waveformOffset + 0] = mono * 0.5f + 0.5f;
+		next.texture[waveformOffset + 1] = next.texture[waveformOffset + 0];
+		next.texture[waveformOffset + 2] = next.monoRms;
+		next.texture[waveformOffset + 3] = 1.0f;
+
+		const float bandPosition = static_cast<float>(x) / static_cast<float>(kAudioTextureWidth - 1);
+		const float scaled = bandPosition * static_cast<float>(next.bands.size() - 1);
+		const unsigned bandA = static_cast<unsigned>(std::floor(scaled));
+		const unsigned bandB = std::min<unsigned>(bandA + 1, static_cast<unsigned>(next.bands.size() - 1));
+		const float t = scaled - static_cast<float>(bandA);
+		const float spectrum = next.bands[bandA] * (1.0f - t) + next.bands[bandB] * t;
+		const std::size_t spectrumOffset = (kAudioTextureWidth + x) * 4;
+		next.texture[spectrumOffset + 0] = spectrum;
+		next.texture[spectrumOffset + 1] = next.bands[0];
+		next.texture[spectrumOffset + 2] = next.bands[1];
+		next.texture[spectrumOffset + 3] = next.bands[2];
+	}
+
+	mCurrent = next;
+	return mCurrent;
+}
--- a/apps/LoopThroughWithOpenGLCompositing/AudioSupport.h
+++ b/apps/LoopThroughWithOpenGLCompositing/AudioSupport.h
@@ -0,0 +1,71 @@
+#pragma once
+
+#include <array>
+#include <cstdint>
+#include <deque>
+#include <mutex>
+#include <vector>
+
+constexpr unsigned kAudioSampleRate = 48000;
+constexpr unsigned kAudioChannelCount = 2;
+constexpr unsigned kAudioTextureWidth = 64;
+constexpr unsigned kAudioTextureHeight = 2;
+
+struct AudioFrameBlock
+{
+	std::vector<int32_t> interleavedSamples;
+
+	std::size_t frameCount() const
+	{
+		return interleavedSamples.size() / kAudioChannelCount;
+	}
+};
+
+struct AudioAnalysisSnapshot
+{
+	std::array<float, 2> rms = { 0.0f, 0.0f };
+	std::array<float, 2> peak = { 0.0f, 0.0f };
+	float monoRms = 0.0f;
+	float monoPeak = 0.0f;
+	std::array<float, 4> bands = { 0.0f, 0.0f, 0.0f, 0.0f };
+	std::array<float, kAudioTextureWidth * kAudioTextureHeight * 4> texture = {};
+};
+
+struct AudioStatusSnapshot
+{
+	bool enabled = false;
+	unsigned bufferedSampleFrames = 0;
+	uint64_t underrunCount = 0;
+	AudioAnalysisSnapshot analysis;
+};
+
+class AudioDelayBuffer
+{
+public:
+	void Reset(unsigned delaySampleFrames);
+	void PushInterleaved(const int32_t* samples, std::size_t sampleFrameCount);
+	AudioFrameBlock Pop(std::size_t sampleFrameCount, bool& underrun);
+	unsigned BufferedSampleFrames() const;
+	uint64_t UnderrunCount() const;
+
+private:
+	mutable std::mutex mMutex;
+	std::deque<int32_t> mSamples;
+	uint64_t mUnderrunCount = 0;
+};
+
+class AudioAnalyzer
+{
+public:
+	void Reset();
+	AudioAnalysisSnapshot Analyze(const AudioFrameBlock& block);
+	const AudioAnalysisSnapshot& Current() const { return mCurrent; }
+
+private:
+	std::deque<float> mMonoHistory;
+	std::array<float, 4> mSmoothedBands = { 0.0f, 0.0f, 0.0f, 0.0f };
+	AudioAnalysisSnapshot mCurrent;
+};
+
+uint64_t AudioSampleTimeForVideoFrame(uint64_t videoFrameIndex, uint64_t frameDuration, uint64_t frameTimescale, uint64_t audioSampleRate = kAudioSampleRate);
+unsigned AudioSamplesForVideoFrame(uint64_t videoFrameIndex, uint64_t frameDuration, uint64_t frameTimescale, uint64_t audioSampleRate = kAudioSampleRate);
--- a/apps/LoopThroughWithOpenGLCompositing/OpenGLComposite.cpp
+++ b/apps/LoopThroughWithOpenGLCompositing/OpenGLComposite.cpp
@@ -44,6 +44,7 @@
 #include "OscServer.h"

 #include <algorithm>
+#include <chrono>
 #include <cstdint>
 #include <cstring>
 #include <cctype>
@@ -51,6 +52,7 @@
 #include <set>
 #include <sstream>
 #include <string>
+#include <thread>
 #include <vector>

 #include <initguid.h>
@@ -60,9 +62,18 @@ DEFINE_GUID(IID_PinnedMemoryAllocator,
 namespace
 {
 constexpr GLuint kDecodedVideoTextureUnit = 1;
-constexpr GLuint kSourceHistoryTextureUnitBase = 2;
+constexpr GLuint kAudioDataTextureUnit = 2;
+constexpr GLuint kSourceHistoryTextureUnitBase = 3;
 constexpr GLuint kPackedVideoTextureUnit = 2;
 constexpr GLuint kGlobalParamsBindingPoint = 0;
+constexpr unsigned kVideoPrerollFrameCount = 5;
+constexpr unsigned kAudioOutputWaterLevelSampleFrames = kAudioSampleRate / 2;
+constexpr unsigned kAudioScheduleChunkSampleFrames = kAudioSampleRate / 100;
+constexpr unsigned kDeckLinkOutputAudioChannelCount = 16;
+
+#ifndef GL_RGBA32F
+#define GL_RGBA32F 0x8814
+#endif
 const char* kVertexShaderSource =
 	"#version 430 core\n"
 	"out vec2 vTexCoord;\n"
@@ -315,8 +326,10 @@ void AppendStd140Vec4(std::vector<unsigned char>& buffer, float x, float y, floa
 OpenGLComposite::OpenGLComposite(HWND hWnd, HDC hDC, HGLRC hRC) :
 	hGLWnd(hWnd), hGLDC(hDC), hGLRC(hRC),
 	mCaptureDelegate(NULL), mPlayoutDelegate(NULL),
-	mDLInput(NULL), mDLOutput(NULL), mDLKeyer(NULL),
+	mDLInput(NULL), mDLOutput(NULL), mDLInputConfiguration(NULL), mDLKeyer(NULL),
 	mPlayoutAllocator(NULL),
+	mTotalPlayoutFrames(0),
+	mAudioOutputSampleTime(0),
 	mInputFrameWidth(0), mInputFrameHeight(0),
 	mOutputFrameWidth(0), mOutputFrameHeight(0),
 	mInputDisplayModeName("1080p59.94"),
@@ -332,6 +345,7 @@ OpenGLComposite::OpenGLComposite(HWND hWnd, HDC hDC, HGLRC hRC) :
 	mLayerTempTexture(0),
 	mFBOTexture(0),
 	mOutputTexture(0),
+	mAudioDataTexture(0),
 	mUnpinnedTextureBuffer(0),
 	mDecodeFrameBuf(0),
 	mLayerTempFrameBuf(0),
@@ -347,6 +361,15 @@ OpenGLComposite::OpenGLComposite(HWND hWnd, HDC hDC, HGLRC hRC) :
 	mGlobalParamsUBOSize(0),
 	mViewWidth(0),
 	mViewHeight(0),
+	mAudioEnabled(false),
+	mAudioOutputEnabled(false),
+	mAudioScheduleEnabled(false),
+	mAudioPrerollEnabled(false),
+	mAudioScheduleSilence(false),
+	mAudioScheduleTone(false),
+	mAudioPrerolling(false),
+	mAudioSchedulerRunning(false),
+	mPlayoutCallbackActive(false),
 	mTemporalHistoryNeedsReset(true)
 {
 	InitializeCriticalSection(&pMutex);
@@ -357,11 +380,22 @@ OpenGLComposite::OpenGLComposite(HWND hWnd, HDC hDC, HGLRC hRC) :

 OpenGLComposite::~OpenGLComposite()
 {
+	mAudioSchedulerRunning.store(false);
+	mAudioPacketQueued.notify_all();
+	if (mAudioSchedulerThread.joinable())
+		mAudioSchedulerThread.join();
+
 	// Cleanup for Capture
 	if (mDLInput != NULL)
 	{
 		mDLInput->SetCallback(NULL);

+		if (mDLInputConfiguration != NULL)
+		{
+			mDLInputConfiguration->Release();
+			mDLInputConfiguration = NULL;
+		}
+
 		mDLInput->Release();
 		mDLInput = NULL;
 	}
@@ -394,6 +428,7 @@ OpenGLComposite::~OpenGLComposite()
 		}

 		mDLOutput->SetScheduledFrameCompletionCallback(NULL);
+		mDLOutput->SetAudioCallback(NULL);

 		mDLOutput->Release();
 		mDLOutput = NULL;
@@ -435,6 +470,8 @@ OpenGLComposite::~OpenGLComposite()
 		glDeleteTextures(1, &mFBOTexture);
 	if (mOutputTexture != 0)
 		glDeleteTextures(1, &mOutputTexture);
+	if (mAudioDataTexture != 0)
+		glDeleteTextures(1, &mAudioDataTexture);
 	if (mOutputFrameBuf != 0)
 		glDeleteFramebuffers(1, &mOutputFrameBuf);
 	if (mUnpinnedTextureBuffer != 0)
@@ -617,6 +654,11 @@ bool OpenGLComposite::InitDeckLink()

 	if (! CheckOpenGLExtensions())
 		goto error;
+	if (mAudioOutputEnabled)
+	{
+		mFastTransferExtensionAvailable = false;
+		OutputDebugStringA("Audio output enabled; using DeckLink-owned output video frames for SDI stability.\n");
+	}
 	if (mInputFrameWidth != mOutputFrameWidth || mInputFrameHeight != mOutputFrameHeight)
 	{
 		mFastTransferExtensionAvailable = false;
@@ -667,6 +709,31 @@ bool OpenGLComposite::InitDeckLink()
 			goto error;
 	}

+	mAudioEnabled = mRuntimeHost ? mRuntimeHost->AudioEnabled() : true;
+	mAudioOutputEnabled = mAudioEnabled && (mRuntimeHost ? mRuntimeHost->AudioOutputEnabled() : true);
+	mAudioScheduleEnabled = mAudioOutputEnabled && (mRuntimeHost ? mRuntimeHost->AudioScheduleEnabled() : true);
+	mAudioPrerollEnabled = mAudioScheduleEnabled && (mRuntimeHost ? mRuntimeHost->AudioPrerollEnabled() : true);
+	mAudioScheduleSilence = mAudioScheduleEnabled && (mRuntimeHost ? mRuntimeHost->AudioScheduleSilence() : false);
+	mAudioScheduleTone = mAudioScheduleEnabled && (mRuntimeHost ? mRuntimeHost->AudioScheduleTone() : false);
+	if (mAudioEnabled)
+	{
+		if (mDLInput->QueryInterface(IID_IDeckLinkConfiguration, (void**)&mDLInputConfiguration) == S_OK && mDLInputConfiguration != NULL)
+		{
+			if (mDLInputConfiguration->SetInt(bmdDeckLinkConfigAudioInputConnection, bmdAudioConnectionEmbedded) != S_OK)
+				OutputDebugStringA("Could not force DeckLink audio input connection to embedded; using current device setting.\n");
+		}
+		else
+		{
+			OutputDebugStringA("Could not query DeckLink input configuration; using current audio input connection.\n");
+		}
+	}
+
+	if (mAudioEnabled && mDLInput->EnableAudioInput(bmdAudioSampleRate48kHz, bmdAudioSampleType32bitInteger, kAudioChannelCount) != S_OK)
+	{
+		OutputDebugStringA("Could not enable DeckLink audio input; continuing without audio.\n");
+		mAudioEnabled = false;
+	}
+
 	mCaptureDelegate = new CaptureDelegate(this);
 	if (mDLInput->SetCallback(mCaptureDelegate) != S_OK)
 		goto error;
@@ -680,6 +747,12 @@ bool OpenGLComposite::InitDeckLink()
 	if (mDLOutput->EnableVideoOutput(outputDisplayMode, bmdVideoOutputFlagDefault) != S_OK)
 		goto error;

+	if (mAudioOutputEnabled && mDLOutput->EnableAudioOutput(bmdAudioSampleRate48kHz, bmdAudioSampleType32bitInteger, kDeckLinkOutputAudioChannelCount, bmdAudioOutputStreamContinuous) != S_OK)
+	{
+		OutputDebugStringA("Could not enable DeckLink audio output; continuing without audio.\n");
+		mAudioOutputEnabled = false;
+	}
+
 	if (mDLOutput->QueryInterface(IID_IDeckLinkKeyer, (void**)&mDLKeyer) == S_OK && mDLKeyer != NULL)
 		mDeckLinkKeyerInterfaceAvailable = true;

@@ -730,6 +803,13 @@ bool OpenGLComposite::InitDeckLink()
 		// If you want RGB 4:4:4 format to be played out "over the wire" in SDI, turn on the "Use 4:4:4 SDI" in the control
 		// panel or turn on the bmdDeckLinkConfig444SDIVideoOutput flag using the IDeckLinkConfiguration interface.
 		IDeckLinkMutableVideoFrame* outputFrame;
+		if (mAudioOutputEnabled)
+		{
+			if (mDLOutput->CreateVideoFrame(mOutputFrameWidth, mOutputFrameHeight, outputFrameRowBytes, bmdFormat8BitBGRA, bmdFrameFlagFlipVertical, &outputFrame) != S_OK)
+				goto error;
+		}
+		else
+		{
 			IDeckLinkVideoBuffer* outputFrameBuffer = NULL;

 			if (mPlayoutAllocator->AllocateVideoBuffer(&outputFrameBuffer) != S_OK)
@@ -737,6 +817,10 @@ bool OpenGLComposite::InitDeckLink()

 			if (mDLOutput->CreateVideoFrameWithBuffer(mOutputFrameWidth, mOutputFrameHeight, outputFrameRowBytes, bmdFormat8BitBGRA, bmdFrameFlagFlipVertical, outputFrameBuffer, &outputFrame) != S_OK)
 				goto error;
+		}
+
+		if (outputFrame == NULL)
+			goto error;

 		mDLOutputVideoFrameQueue.push_back(outputFrame);
 	}
@@ -748,6 +832,13 @@ bool OpenGLComposite::InitDeckLink()
 	if (mDLOutput->SetScheduledFrameCompletionCallback(mPlayoutDelegate) != S_OK)
 		goto error;

+	if (mAudioOutputEnabled && mDLOutput->SetAudioCallback(mPlayoutDelegate) != S_OK)
+	{
+		OutputDebugStringA("Could not set DeckLink audio output callback; continuing without audio.\n");
+		mDLOutput->DisableAudioOutput();
+		mAudioOutputEnabled = false;
+	}
+
 	bSuccess = true;

 error:
@@ -770,6 +861,11 @@ error:
 			mDLOutput->Release();
 			mDLOutput = NULL;
 		}
+		if (mDLInputConfiguration != NULL)
+		{
+			mDLInputConfiguration->Release();
+			mDLInputConfiguration = NULL;
+		}
 	}

 	if (pDL != NULL)
@@ -1052,6 +1148,14 @@ bool OpenGLComposite::InitOpenGLState()
 	glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
 	glTexImage2D(GL_TEXTURE_2D, 0, GL_RGBA8, mOutputFrameWidth, mOutputFrameHeight, 0, GL_BGRA, GL_UNSIGNED_INT_8_8_8_8_REV, NULL);

+	glGenTextures(1, &mAudioDataTexture);
+	glBindTexture(GL_TEXTURE_2D, mAudioDataTexture);
+	glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_LINEAR);
+	glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
+	glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
+	glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
+	glTexImage2D(GL_TEXTURE_2D, 0, GL_RGBA32F, kAudioTextureWidth, kAudioTextureHeight, 0, GL_RGBA, GL_FLOAT, mAudioAnalysis.texture.data());
+
 	glBindFramebuffer(GL_FRAMEBUFFER, mOutputFrameBuf);
 	glFramebufferTexture2D(GL_FRAMEBUFFER, GL_COLOR_ATTACHMENT0, GL_TEXTURE_2D, mOutputTexture, 0);
 	glStatus = glCheckFramebufferStatus(GL_FRAMEBUFFER);
@@ -1135,16 +1239,217 @@ void OpenGLComposite::VideoFrameArrived(IDeckLinkVideoInputFrame* inputFrame, bo
 	inputFrameBuffer->Release();
 }

+void OpenGLComposite::AudioPacketArrived(IDeckLinkAudioInputPacket* audioPacket)
+{
+	if (!mAudioEnabled || !audioPacket)
+		return;
+
+	void* audioBytes = nullptr;
+	if (audioPacket->GetBytes(&audioBytes) != S_OK || !audioBytes)
+		return;
+
+	const long sampleFrameCount = audioPacket->GetSampleFrameCount();
+	if (sampleFrameCount <= 0)
+		return;
+
+	TimestampedAudioPacket packet;
+	packet.block.interleavedSamples.assign(
+		static_cast<const int32_t*>(audioBytes),
+		static_cast<const int32_t*>(audioBytes) + (static_cast<std::size_t>(sampleFrameCount) * kAudioChannelCount));
+
+	if (!mAudioScheduleEnabled)
+	{
+		AudioAnalysisSnapshot audioAnalysis;
+		{
+			std::lock_guard<std::mutex> analyzerLock(mAudioAnalyzerMutex);
+			audioAnalysis = mAudioAnalyzer.Analyze(packet.block);
+		}
+		{
+			std::lock_guard<std::mutex> audioLock(mAudioStateMutex);
+			mAudioAnalysis = audioAnalysis;
+		}
+		updateAudioStatus();
+		return;
+	}
+
+	{
+		std::lock_guard<std::mutex> audioLock(mAudioStateMutex);
+		for (int32_t sample : packet.block.interleavedSamples)
+			mAudioSampleQueue.push_back(sample);
+		mQueuedAudioSampleFrames += static_cast<unsigned>(sampleFrameCount);
+	}
+	mAudioPacketQueued.notify_one();
+	ScheduleAudioToWaterLevel();
+}
+
+HRESULT OpenGLComposite::RenderAudioSamples(BOOL preroll)
+{
+	return ScheduleAudioToWaterLevel();
+}
+
+HRESULT OpenGLComposite::ScheduleAudioToWaterLevel()
+{
+	if (!mAudioScheduleEnabled || !mDLOutput)
+		return S_OK;
+
+	if (mPlayoutCallbackActive.load(std::memory_order_acquire))
+		return S_FALSE;
+
+	unsigned bufferedSampleFrames = 0;
+	if (mDLOutput->GetBufferedAudioSampleFrameCount(&bufferedSampleFrames) != S_OK)
+	{
+		OutputDebugStringA("Could not query DeckLink buffered audio sample count.\n");
+		return E_FAIL;
+	}
+
+	const unsigned audioWaterLevel = static_cast<unsigned>(AudioSampleTimeForVideoFrame(kVideoPrerollFrameCount, mFrameDuration, mFrameTimescale));
+	if (bufferedSampleFrames >= audioWaterLevel)
+		return S_OK;
+
+	TimestampedAudioPacket packet;
+	bool poppedCapturedAudio = false;
+	{
+		std::unique_lock<std::mutex> audioLock(mAudioStateMutex, std::try_to_lock);
+		if (!audioLock.owns_lock())
+			return S_FALSE;
+
+		const unsigned audioDeficitFrames = audioWaterLevel - bufferedSampleFrames;
+		const unsigned requestedFrames = audioDeficitFrames < kAudioScheduleChunkSampleFrames ? audioDeficitFrames : kAudioScheduleChunkSampleFrames;
+		if (requestedFrames == 0)
+			return S_OK;
+
+		if (mAudioScheduleTone)
+		{
+			const std::size_t requestedSamples = static_cast<std::size_t>(requestedFrames) * kAudioChannelCount;
+			packet.block.interleavedSamples.reserve(requestedSamples);
+			for (unsigned frame = 0; frame < requestedFrames; ++frame)
+			{
+				const double phase = (static_cast<double>(mAudioToneSampleIndex++) * 440.0 * 6.28318530717958647692) / static_cast<double>(kAudioSampleRate);
+				const int32_t sample = static_cast<int32_t>(std::sin(phase) * 0.125 * 2147483647.0);
+				for (unsigned channel = 0; channel < kAudioChannelCount; ++channel)
+					packet.block.interleavedSamples.push_back(sample);
+			}
+		}
+		else if (mAudioScheduleSilence)
+		{
+			packet.block.interleavedSamples.assign(static_cast<std::size_t>(requestedFrames) * kAudioChannelCount, 0);
+		}
+		else
+		{
+			const std::size_t requestedSamples = static_cast<std::size_t>(requestedFrames) * kAudioChannelCount;
+			packet.block.interleavedSamples.reserve(requestedSamples);
+			while (!mAudioSampleQueue.empty() && packet.block.interleavedSamples.size() < requestedSamples)
+			{
+				packet.block.interleavedSamples.push_back(mAudioSampleQueue.front());
+				mAudioSampleQueue.pop_front();
+			}
+			if (packet.block.interleavedSamples.size() < requestedSamples)
+			{
+				mAudioUnderrunCount++;
+				packet.block.interleavedSamples.resize(requestedSamples, 0);
+			}
+			const auto frameCount = static_cast<unsigned>(packet.block.frameCount());
+			mQueuedAudioSampleFrames = frameCount <= mQueuedAudioSampleFrames ? mQueuedAudioSampleFrames - frameCount : 0;
+			poppedCapturedAudio = true;
+		}
+	}
+
+	const unsigned sampleFrames = static_cast<unsigned>(packet.block.frameCount());
+	if (sampleFrames == 0)
+		return S_FALSE;
+
+	std::vector<int32_t> deckLinkAudioSamples(static_cast<std::size_t>(sampleFrames) * kDeckLinkOutputAudioChannelCount, 0);
+	for (unsigned frame = 0; frame < sampleFrames; ++frame)
+	{
+		const std::size_t source = static_cast<std::size_t>(frame) * kAudioChannelCount;
+		const std::size_t destination = static_cast<std::size_t>(frame) * kDeckLinkOutputAudioChannelCount;
+		deckLinkAudioSamples[destination] = packet.block.interleavedSamples[source];
+		deckLinkAudioSamples[destination + 1] = packet.block.interleavedSamples[source + 1];
+	}
+
+	if (mPlayoutCallbackActive.load(std::memory_order_acquire))
+	{
+		std::lock_guard<std::mutex> audioLock(mAudioStateMutex);
+		if (poppedCapturedAudio)
+		{
+			for (auto it = packet.block.interleavedSamples.rbegin(); it != packet.block.interleavedSamples.rend(); ++it)
+				mAudioSampleQueue.push_front(*it);
+			mQueuedAudioSampleFrames += sampleFrames;
+		}
+		return S_FALSE;
+	}
+
+	unsigned sampleFramesWritten = 0;
+	HRESULT scheduleResult = mDLOutput->ScheduleAudioSamples(
+		deckLinkAudioSamples.data(),
+		sampleFrames,
+		static_cast<BMDTimeValue>(mAudioOutputSampleTime),
+		kAudioSampleRate,
+		&sampleFramesWritten);
+
+	if (scheduleResult == S_OK)
+	{
+		if (sampleFramesWritten == 0 && sampleFrames > 0)
+			OutputDebugStringA("DeckLink accepted audio schedule call but wrote 0 sample frames.\n");
+		mAudioOutputSampleTime += sampleFramesWritten;
+
+		AudioFrameBlock analysisBlock = packet.block;
+		AudioAnalysisSnapshot audioAnalysis;
+		{
+			std::lock_guard<std::mutex> analyzerLock(mAudioAnalyzerMutex);
+			audioAnalysis = mAudioAnalyzer.Analyze(analysisBlock);
+		}
+		{
+			std::lock_guard<std::mutex> audioLock(mAudioStateMutex);
+			mAudioAnalysis = audioAnalysis;
+			packet.scheduledOutputSamples = std::move(deckLinkAudioSamples);
+			mScheduledAudioPacketRetainQueue.push_back(std::move(packet));
+			while (mScheduledAudioPacketRetainQueue.size() > 64)
+				mScheduledAudioPacketRetainQueue.pop_front();
+		}
+		updateAudioStatus();
+	}
+	else
+	{
+		OutputDebugStringA("DeckLink ScheduleAudioSamples failed while topping up audio output.\n");
+	}
+
+	return scheduleResult;
+}
+
+void OpenGLComposite::AudioSchedulingLoop()
+{
+	while (mAudioSchedulerRunning.load())
+	{
+		ScheduleAudioToWaterLevel();
+
+		std::unique_lock<std::mutex> audioLock(mAudioStateMutex);
+		mAudioPacketQueued.wait_for(audioLock, std::chrono::milliseconds(20), [this]()
+		{
+			return !mAudioSchedulerRunning.load() || !mAudioPacketQueue.empty();
+		});
+	}
+}
+
 // Render the live video texture through the runtime shader into the off-screen framebuffer.
 // Read the result back from the frame buffer and schedule it for playout.
 void OpenGLComposite::PlayoutFrameCompleted(IDeckLinkVideoFrame* completedFrame, BMDOutputFrameCompletionResult completionResult)
 {
+	mPlayoutCallbackActive.store(true, std::memory_order_release);
 	EnterCriticalSection(&pMutex);
+	auto leavePlayoutCallback = [this]()
+	{
+		mPlayoutCallbackActive.store(false, std::memory_order_release);
+		LeaveCriticalSection(&pMutex);
+	};

-	// Get the first frame from the queue
-	IDeckLinkMutableVideoFrame* outputVideoFrame = mDLOutputVideoFrameQueue.front();
-	mDLOutputVideoFrameQueue.push_back(outputVideoFrame);
-	mDLOutputVideoFrameQueue.pop_front();
+	if (!completedFrame)
+	{
+		leavePlayoutCallback();
+		return;
+	}
+
+	IDeckLinkVideoFrame* outputVideoFrame = completedFrame;

 	// make GL context current in this thread
 	wglMakeCurrent( hGLDC, hGLRC );
@@ -1177,14 +1482,14 @@ void OpenGLComposite::PlayoutFrameCompleted(IDeckLinkVideoFrame* completedFrame,
 	IDeckLinkVideoBuffer* outputVideoFrameBuffer;
 	if (outputVideoFrame->QueryInterface(IID_IDeckLinkVideoBuffer, (void**)&outputVideoFrameBuffer) != S_OK)
 	{
-		LeaveCriticalSection(&pMutex);
+		leavePlayoutCallback();
 		return;
 	}

 	if (outputVideoFrameBuffer->StartAccess(bmdBufferAccessWrite) != S_OK)
 	{
 		outputVideoFrameBuffer->Release();
-		LeaveCriticalSection(&pMutex);
+		leavePlayoutCallback();
 		return;
 	}

@@ -1225,15 +1530,30 @@ void OpenGLComposite::PlayoutFrameCompleted(IDeckLinkVideoFrame* completedFrame,

 	wglMakeCurrent( NULL, NULL );

-	LeaveCriticalSection(&pMutex);
+	leavePlayoutCallback();
 }

 bool OpenGLComposite::Start()
 {
 	mTotalPlayoutFrames = 0;
+	initializeAudioDelay();
+	if (mAudioPrerollEnabled)
+	{
+		mDLOutput->FlushBufferedAudioSamples();
+		if (mDLOutput->BeginAudioPreroll() != S_OK)
+		{
+			OutputDebugStringA("Could not begin DeckLink audio preroll; continuing without audio.\n");
+			mDLOutput->DisableAudioOutput();
+			mAudioOutputEnabled = false;
+		}
+		else
+		{
+			mAudioPrerolling = true;
+		}
+	}

 	// Preroll frames
-	for (unsigned i = 0; i < 5; i++)
+	for (unsigned i = 0; i < kVideoPrerollFrameCount; i++)
 	{
 		// Take each video frame from the front of the queue and move it to the back
 		IDeckLinkMutableVideoFrame* outputVideoFrame = mDLOutputVideoFrameQueue.front();
@@ -1264,8 +1584,37 @@ bool OpenGLComposite::Start()
 		mTotalPlayoutFrames++;
 	}

-	mDLInput->StartStreams();
-	mDLOutput->StartScheduledPlayback(0, mFrameTimescale, 1.0);
+	if (mDLInput->StartStreams() != S_OK)
+	{
+		return false;
+	}
+
+	if (mAudioPrerolling)
+	{
+		const unsigned audioWaterLevel = static_cast<unsigned>(AudioSampleTimeForVideoFrame(kVideoPrerollFrameCount, mFrameDuration, mFrameTimescale));
+		const auto prerollDeadline = std::chrono::steady_clock::now() + std::chrono::milliseconds(750);
+		while (mAudioScheduleEnabled && std::chrono::steady_clock::now() < prerollDeadline)
+		{
+			unsigned bufferedSampleFrames = 0;
+			if (mDLOutput->GetBufferedAudioSampleFrameCount(&bufferedSampleFrames) == S_OK && bufferedSampleFrames >= audioWaterLevel)
+				break;
+			std::this_thread::sleep_for(std::chrono::milliseconds(5));
+		}
+
+		if (mDLOutput->EndAudioPreroll() != S_OK)
+		{
+			OutputDebugStringA("Could not end DeckLink audio preroll; continuing without audio.\n");
+			mDLOutput->DisableAudioOutput();
+			mAudioOutputEnabled = false;
+			mAudioScheduleEnabled = false;
+		}
+		mAudioPrerolling = false;
+	}
+
+	if (mDLOutput->StartScheduledPlayback(0, mFrameTimescale, 1.0) != S_OK)
+	{
+		return false;
+	}

 	return true;
 }
@@ -1295,11 +1644,23 @@ bool OpenGLComposite::Stop()
 		}
 	}

+	mAudioSchedulerRunning.store(false);
+	mAudioPacketQueued.notify_all();
+	if (mAudioSchedulerThread.joinable())
+		mAudioSchedulerThread.join();
+
 	mDLInput->StopStreams();
 	mDLInput->DisableVideoInput();
+	if (mAudioEnabled)
+		mDLInput->DisableAudioInput();

 	mDLOutput->StopScheduledPlayback(0, NULL, 0);
+	mDLOutput->SetAudioCallback(NULL);
+	mDLOutput->SetScheduledFrameCompletionCallback(NULL);
 	mDLOutput->DisableVideoOutput();
+	mAudioPrerolling = false;
+	if (mAudioOutputEnabled)
+		mDLOutput->DisableAudioOutput();

 	return true;
 }
@@ -1411,6 +1772,9 @@ bool OpenGLComposite::compileSingleLayerProgram(const RuntimeRenderState& state,
 	const GLint videoInputLocation = glGetUniformLocation(newProgram.get(), "gVideoInput");
 	if (videoInputLocation >= 0)
 		glUniform1i(videoInputLocation, static_cast<GLint>(kDecodedVideoTextureUnit));
+	const GLint audioDataLocation = glGetUniformLocation(newProgram.get(), "gAudioData");
+	if (audioDataLocation >= 0)
+		glUniform1i(audioDataLocation, static_cast<GLint>(kAudioDataTextureUnit));
 	for (unsigned index = 0; index < historyCap; ++index)
 	{
 		const std::string sourceSamplerName = "gSourceHistory" + std::to_string(index);
@@ -1973,6 +2337,8 @@ void OpenGLComposite::renderShaderProgram(GLuint sourceTexture, GLuint destinati
 	glClear(GL_COLOR_BUFFER_BIT | GL_DEPTH_BUFFER_BIT);
 	glActiveTexture(GL_TEXTURE0 + kDecodedVideoTextureUnit);
 	glBindTexture(GL_TEXTURE_2D, sourceTexture);
+	glActiveTexture(GL_TEXTURE0 + kAudioDataTextureUnit);
+	glBindTexture(GL_TEXTURE_2D, mAudioDataTexture);
 	bindHistorySamplers(state, sourceTexture);
 	bindLayerTextureAssets(layerProgram);
 	glBindVertexArray(mFullscreenVAO);
@@ -1995,6 +2361,8 @@ void OpenGLComposite::renderShaderProgram(GLuint sourceTexture, GLuint destinati
 		glActiveTexture(GL_TEXTURE0 + shaderTextureBase + static_cast<GLuint>(index));
 		glBindTexture(GL_TEXTURE_2D, 0);
 	}
+	glActiveTexture(GL_TEXTURE0 + kAudioDataTextureUnit);
+	glBindTexture(GL_TEXTURE_2D, 0);
 	glActiveTexture(GL_TEXTURE0 + kDecodedVideoTextureUnit);
 	glBindTexture(GL_TEXTURE_2D, 0);
 	glActiveTexture(GL_TEXTURE0);
@@ -2066,6 +2434,58 @@ void OpenGLComposite::broadcastRuntimeState()
 		mControlServer->BroadcastState();
 }

+BMDTimeValue OpenGLComposite::delayedAudioStreamTime() const
+{
+	return static_cast<BMDTimeValue>(kVideoPrerollFrameCount) * mFrameDuration;
+}
+
+void OpenGLComposite::initializeAudioDelay()
+{
+	{
+		std::lock_guard<std::mutex> analyzerLock(mAudioAnalyzerMutex);
+		mAudioAnalyzer.Reset();
+	}
+	{
+		std::lock_guard<std::mutex> audioLock(mAudioStateMutex);
+		mAudioAnalysis = AudioAnalysisSnapshot();
+		mAudioPacketQueue.clear();
+		mScheduledAudioPacketRetainQueue.clear();
+		mAudioSampleQueue.clear();
+		mQueuedAudioSampleFrames = 0;
+		mAudioUnderrunCount = 0;
+		mAudioOutputSampleTime = 0;
+		mAudioToneSampleIndex = 0;
+		mHasFirstAudioPacketTime = false;
+		mFirstAudioPacketTime = 0;
+	}
+	updateAudioStatus();
+}
+
+void OpenGLComposite::updateAudioDataTexture(const AudioAnalysisSnapshot& analysis)
+{
+	if (mAudioDataTexture == 0)
+		return;
+
+	glActiveTexture(GL_TEXTURE0 + kAudioDataTextureUnit);
+	glBindTexture(GL_TEXTURE_2D, mAudioDataTexture);
+	glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, kAudioTextureWidth, kAudioTextureHeight, GL_RGBA, GL_FLOAT, analysis.texture.data());
+	glBindTexture(GL_TEXTURE_2D, 0);
+	glActiveTexture(GL_TEXTURE0);
+}
+
+void OpenGLComposite::updateAudioStatus()
+{
+	if (!mRuntimeHost)
+		return;
+
+	AudioStatusSnapshot status;
+	status.enabled = mAudioEnabled;
+	status.bufferedSampleFrames = mQueuedAudioSampleFrames;
+	status.underrunCount = mAudioUnderrunCount;
+	status.analysis = mAudioAnalysis;
+	mRuntimeHost->SetAudioStatus(status);
+}
+
 bool OpenGLComposite::updateGlobalParamsBuffer(const RuntimeRenderState& state, unsigned availableSourceHistoryLength, unsigned availableTemporalHistoryLength)
 {
 	std::vector<unsigned char> buffer;
@@ -2085,6 +2505,15 @@ bool OpenGLComposite::updateGlobalParamsBuffer(const RuntimeRenderState& state,
 		: 0u;
 	AppendStd140Int(buffer, static_cast<int>(effectiveSourceHistoryLength));
 	AppendStd140Int(buffer, static_cast<int>(effectiveTemporalHistoryLength));
+	AppendStd140Vec2(buffer, state.audioAnalysis.rms[0], state.audioAnalysis.rms[1]);
+	AppendStd140Vec2(buffer, state.audioAnalysis.peak[0], state.audioAnalysis.peak[1]);
+	AppendStd140Float(buffer, state.audioAnalysis.monoRms);
+	AppendStd140Float(buffer, state.audioAnalysis.monoPeak);
+	AppendStd140Vec4(buffer,
+		state.audioAnalysis.bands[0],
+		state.audioAnalysis.bands[1],
+		state.audioAnalysis.bands[2],
+		state.audioAnalysis.bands[3]);

 	for (const ShaderParameterDefinition& definition : state.parameterDefinitions)
 	{
@@ -2623,11 +3052,14 @@ ULONG	CaptureDelegate::Release()
 	return newCount;
 }

-HRESULT	CaptureDelegate::VideoInputFrameArrived(IDeckLinkVideoInputFrame* inputFrame, IDeckLinkAudioInputPacket* /*audioPacket*/)
+HRESULT	CaptureDelegate::VideoInputFrameArrived(IDeckLinkVideoInputFrame* inputFrame, IDeckLinkAudioInputPacket* audioPacket)
 {
+	if (audioPacket)
+		m_pOwner->AudioPacketArrived(audioPacket);
+
 	if (! inputFrame)
 	{
-		// It's possible to receive a NULL inputFrame, but a valid audioPacket. Ignore audio-only frame.
+		// It's possible to receive a NULL inputFrame, but a valid audioPacket.
 		return S_OK;
 	}

@@ -2653,6 +3085,23 @@ PlayoutDelegate::PlayoutDelegate(OpenGLComposite* pOwner) :

 HRESULT	PlayoutDelegate::QueryInterface(REFIID iid, LPVOID *ppv)
 {
+	if (ppv == nullptr)
+		return E_POINTER;
+
+	if (iid == IID_IUnknown || iid == IID_IDeckLinkVideoOutputCallback)
+	{
+		*ppv = static_cast<IDeckLinkVideoOutputCallback*>(this);
+		AddRef();
+		return S_OK;
+	}
+
+	if (iid == IID_IDeckLinkAudioOutputCallback)
+	{
+		*ppv = static_cast<IDeckLinkAudioOutputCallback*>(this);
+		AddRef();
+		return S_OK;
+	}
+
 	*ppv = NULL;
 	return E_NOINTERFACE;
 }
@@ -2694,3 +3143,8 @@ HRESULT	PlayoutDelegate::ScheduledPlaybackHasStopped ()
 {
 	return S_OK;
 }
+
+HRESULT	PlayoutDelegate::RenderAudioSamples (BOOL preroll)
+{
+	return m_pOwner->RenderAudioSamples(preroll);
+}
--- a/apps/LoopThroughWithOpenGLCompositing/OpenGLComposite.h
+++ b/apps/LoopThroughWithOpenGLCompositing/OpenGLComposite.h
@@ -52,13 +52,18 @@
 #include <comutil.h>
 #include "DeckLinkAPI_h.h"

+#include "AudioSupport.h"
 #include "VideoFrameTransfer.h"
 #include "RuntimeHost.h"

 #include <atomic>
+#include <condition_variable>
+#include <cstdint>
 #include <functional>
 #include <map>
 #include <memory>
+#include <mutex>
+#include <thread>
 #include <vector>
 #include <deque>

@@ -96,6 +101,10 @@ public:
 	void paintGL();

 	void VideoFrameArrived(IDeckLinkVideoInputFrame* inputFrame, bool hasNoInputSource);
+	void AudioPacketArrived(IDeckLinkAudioInputPacket* audioPacket);
+	HRESULT RenderAudioSamples(BOOL preroll);
+	HRESULT ScheduleAudioToWaterLevel();
+	void AudioSchedulingLoop();
 	void PlayoutFrameCompleted(IDeckLinkVideoFrame* completedFrame, BMDOutputFrameCompletionResult result);

 private:
@@ -112,12 +121,14 @@ private:
 	// DeckLink
 	IDeckLinkInput*							mDLInput;
 	IDeckLinkOutput*						mDLOutput;
+	IDeckLinkConfiguration*					mDLInputConfiguration;
 	IDeckLinkKeyer*							mDLKeyer;
 	std::deque<IDeckLinkMutableVideoFrame*>	mDLOutputVideoFrameQueue;
 	PinnedMemoryAllocator*					mPlayoutAllocator;
 	BMDTimeValue							mFrameDuration;
 	BMDTimeScale							mFrameTimescale;
 	unsigned								mTotalPlayoutFrames;
+	uint64_t								mAudioOutputSampleTime;
 	unsigned								mInputFrameWidth;
 	unsigned								mInputFrameHeight;
 	unsigned								mOutputFrameWidth;
@@ -139,6 +150,7 @@ private:
 	GLuint									mLayerTempTexture;
 	GLuint									mFBOTexture;
 	GLuint									mOutputTexture;
+	GLuint									mAudioDataTexture;
 	GLuint									mUnpinnedTextureBuffer;
 	GLuint									mDecodeFrameBuf;
 	GLuint									mLayerTempFrameBuf;
@@ -157,6 +169,35 @@ private:
 	std::unique_ptr<RuntimeHost>				mRuntimeHost;
 	std::unique_ptr<ControlServer>			mControlServer;
 	std::unique_ptr<OscServer>				mOscServer;
+	bool									mAudioEnabled;
+	bool									mAudioOutputEnabled;
+	bool									mAudioScheduleEnabled;
+	bool									mAudioPrerollEnabled;
+	bool									mAudioScheduleSilence;
+	bool									mAudioScheduleTone;
+	bool									mAudioPrerolling;
+	std::atomic<bool>						mAudioSchedulerRunning;
+	std::atomic<bool>						mPlayoutCallbackActive;
+	std::thread								mAudioSchedulerThread;
+	std::mutex								mAudioStateMutex;
+	std::mutex								mAudioAnalyzerMutex;
+	AudioAnalyzer							mAudioAnalyzer;
+	AudioAnalysisSnapshot					mAudioAnalysis;
+	struct TimestampedAudioPacket
+	{
+		AudioFrameBlock block;
+		std::vector<int32_t> scheduledOutputSamples;
+		BMDTimeValue streamTime = 0;
+	};
+	std::deque<TimestampedAudioPacket>		mAudioPacketQueue;
+	std::deque<TimestampedAudioPacket>		mScheduledAudioPacketRetainQueue;
+	std::deque<int32_t>						mAudioSampleQueue;
+	std::condition_variable					mAudioPacketQueued;
+	unsigned								mQueuedAudioSampleFrames = 0;
+	uint64_t								mAudioUnderrunCount = 0;
+	uint64_t								mAudioToneSampleIndex = 0;
+	bool									mHasFirstAudioPacketTime = false;
+	BMDTimeValue							mFirstAudioPacketTime = 0;

 	struct LayerProgram
 	{
@@ -209,6 +250,10 @@ private:
 	void renderEffect();
 	bool PollRuntimeChanges();
 	void broadcastRuntimeState();
+	void initializeAudioDelay();
+	BMDTimeValue delayedAudioStreamTime() const;
+	void updateAudioDataTexture(const AudioAnalysisSnapshot& analysis);
+	void updateAudioStatus();
 	bool updateGlobalParamsBuffer(const RuntimeRenderState& state, unsigned availableSourceHistoryLength, unsigned availableTemporalHistoryLength);
 	bool validateTemporalTextureUnitBudget(const std::vector<RuntimeRenderState>& layerStates, std::string& error) const;
 	bool ensureTemporalHistoryResources(const std::vector<RuntimeRenderState>& layerStates, std::string& error);
@@ -341,7 +386,7 @@ public:
 // Render Delegate Class
 ////////////////////////////////////////////

-class PlayoutDelegate : public IDeckLinkVideoOutputCallback
+class PlayoutDelegate : public IDeckLinkVideoOutputCallback, public IDeckLinkAudioOutputCallback
 {
 	OpenGLComposite*	m_pOwner;
 	LONG				mRefCount;
@@ -356,6 +401,7 @@ public:

 	virtual HRESULT	STDMETHODCALLTYPE	ScheduledFrameCompleted (IDeckLinkVideoFrame* completedFrame, BMDOutputFrameCompletionResult result);
 	virtual HRESULT	STDMETHODCALLTYPE	ScheduledPlaybackHasStopped ();
+	virtual HRESULT	STDMETHODCALLTYPE	RenderAudioSamples (BOOL preroll);
 };

 #endif		// __OPENGL_COMPOSITE_H__
--- a/apps/LoopThroughWithOpenGLCompositing/RuntimeHost.cpp
+++ b/apps/LoopThroughWithOpenGLCompositing/RuntimeHost.cpp
@@ -1055,6 +1055,12 @@ void RuntimeHost::SetPerformanceStats(double frameBudgetMilliseconds, double ren
 		mSmoothedRenderMilliseconds = mSmoothedRenderMilliseconds * 0.9 + mRenderMilliseconds * 0.1;
 }

+void RuntimeHost::SetAudioStatus(const AudioStatusSnapshot& status)
+{
+	std::lock_guard<std::mutex> lock(mMutex);
+	mAudioStatus = status;
+}
+
 void RuntimeHost::AdvanceFrame()
 {
 	std::lock_guard<std::mutex> lock(mMutex);
@@ -1121,6 +1127,7 @@ std::vector<RuntimeRenderState> RuntimeHost::GetLayerRenderStates(unsigned outpu
 		state.inputHeight = mSignalHeight;
 		state.outputWidth = outputWidth;
 		state.outputHeight = outputHeight;
+		state.audioAnalysis = mAudioStatus.analysis;
 		state.parameterDefinitions = shaderIt->second.parameters;
 		state.textureAssets = shaderIt->second.textureAssets;
 		state.isTemporal = shaderIt->second.temporal.enabled;
@@ -1182,6 +1189,31 @@ bool RuntimeHost::LoadConfig(std::string& error)
 	}
 	if (const JsonValue* enableExternalKeyingValue = configJson.find("enableExternalKeying"))
 		mConfig.enableExternalKeying = enableExternalKeyingValue->asBoolean(mConfig.enableExternalKeying);
+	if (const JsonValue* audioEnabledValue = configJson.find("audioEnabled"))
+		mConfig.audioEnabled = audioEnabledValue->asBoolean(mConfig.audioEnabled);
+	if (const JsonValue* audioOutputEnabledValue = configJson.find("audioOutputEnabled"))
+		mConfig.audioOutputEnabled = audioOutputEnabledValue->asBoolean(mConfig.audioOutputEnabled);
+	if (const JsonValue* audioScheduleEnabledValue = configJson.find("audioScheduleEnabled"))
+		mConfig.audioScheduleEnabled = audioScheduleEnabledValue->asBoolean(mConfig.audioScheduleEnabled);
+	if (const JsonValue* audioPrerollEnabledValue = configJson.find("audioPrerollEnabled"))
+		mConfig.audioPrerollEnabled = audioPrerollEnabledValue->asBoolean(mConfig.audioPrerollEnabled);
+	if (const JsonValue* audioScheduleSilenceValue = configJson.find("audioScheduleSilence"))
+		mConfig.audioScheduleSilence = audioScheduleSilenceValue->asBoolean(mConfig.audioScheduleSilence);
+	if (const JsonValue* audioScheduleToneValue = configJson.find("audioScheduleTone"))
+		mConfig.audioScheduleTone = audioScheduleToneValue->asBoolean(mConfig.audioScheduleTone);
+	if (const JsonValue* audioChannelCountValue = configJson.find("audioChannelCount"))
+		mConfig.audioChannelCount = static_cast<unsigned>(audioChannelCountValue->asNumber(static_cast<double>(mConfig.audioChannelCount)));
+	if (const JsonValue* audioSampleRateValue = configJson.find("audioSampleRate"))
+		mConfig.audioSampleRate = static_cast<unsigned>(audioSampleRateValue->asNumber(static_cast<double>(mConfig.audioSampleRate)));
+	if (const JsonValue* audioDelayModeValue = configJson.find("audioDelayMode"))
+	{
+		if (audioDelayModeValue->isString() && !audioDelayModeValue->asString().empty())
+			mConfig.audioDelayMode = audioDelayModeValue->asString();
+	}
+	if (mConfig.audioChannelCount != kAudioChannelCount)
+		mConfig.audioChannelCount = kAudioChannelCount;
+	if (mConfig.audioSampleRate != kAudioSampleRate)
+		mConfig.audioSampleRate = kAudioSampleRate;
 	if (const JsonValue* videoFormatValue = configJson.find("videoFormat"))
 	{
 		if (videoFormatValue->isString() && !videoFormatValue->asString().empty())
@@ -1519,6 +1551,15 @@ JsonValue RuntimeHost::BuildStateValue() const
 	app.set("autoReload", JsonValue(mAutoReloadEnabled));
 	app.set("maxTemporalHistoryFrames", JsonValue(static_cast<double>(mConfig.maxTemporalHistoryFrames)));
 	app.set("enableExternalKeying", JsonValue(mConfig.enableExternalKeying));
+	app.set("audioEnabled", JsonValue(mConfig.audioEnabled));
+	app.set("audioOutputEnabled", JsonValue(mConfig.audioOutputEnabled));
+	app.set("audioScheduleEnabled", JsonValue(mConfig.audioScheduleEnabled));
+	app.set("audioPrerollEnabled", JsonValue(mConfig.audioPrerollEnabled));
+	app.set("audioScheduleSilence", JsonValue(mConfig.audioScheduleSilence));
+	app.set("audioScheduleTone", JsonValue(mConfig.audioScheduleTone));
+	app.set("audioChannelCount", JsonValue(static_cast<double>(mConfig.audioChannelCount)));
+	app.set("audioSampleRate", JsonValue(static_cast<double>(mConfig.audioSampleRate)));
+	app.set("audioDelayMode", JsonValue(mConfig.audioDelayMode));
 	app.set("inputVideoFormat", JsonValue(mConfig.inputVideoFormat));
 	app.set("inputFrameRate", JsonValue(mConfig.inputFrameRate));
 	app.set("outputVideoFormat", JsonValue(mConfig.outputVideoFormat));
@@ -1538,6 +1579,26 @@ JsonValue RuntimeHost::BuildStateValue() const
 	video.set("modeName", JsonValue(mSignalModeName));
 	root.set("video", video);

+	JsonValue audio = JsonValue::MakeObject();
+	audio.set("enabled", JsonValue(mAudioStatus.enabled));
+	audio.set("bufferedSampleFrames", JsonValue(static_cast<double>(mAudioStatus.bufferedSampleFrames)));
+	audio.set("underrunCount", JsonValue(static_cast<double>(mAudioStatus.underrunCount)));
+	JsonValue rms = JsonValue::MakeArray();
+	rms.pushBack(JsonValue(static_cast<double>(mAudioStatus.analysis.rms[0])));
+	rms.pushBack(JsonValue(static_cast<double>(mAudioStatus.analysis.rms[1])));
+	audio.set("rms", rms);
+	JsonValue peak = JsonValue::MakeArray();
+	peak.pushBack(JsonValue(static_cast<double>(mAudioStatus.analysis.peak[0])));
+	peak.pushBack(JsonValue(static_cast<double>(mAudioStatus.analysis.peak[1])));
+	audio.set("peak", peak);
+	audio.set("monoRms", JsonValue(static_cast<double>(mAudioStatus.analysis.monoRms)));
+	audio.set("monoPeak", JsonValue(static_cast<double>(mAudioStatus.analysis.monoPeak)));
+	JsonValue bands = JsonValue::MakeArray();
+	for (float band : mAudioStatus.analysis.bands)
+		bands.pushBack(JsonValue(static_cast<double>(band)));
+	audio.set("bands", bands);
+	root.set("audio", audio);
+
 	JsonValue deckLink = JsonValue::MakeObject();
 	deckLink.set("modelName", JsonValue(mDeckLinkOutputStatus.modelName));
 	deckLink.set("supportsInternalKeying", JsonValue(mDeckLinkOutputStatus.supportsInternalKeying));
--- a/apps/LoopThroughWithOpenGLCompositing/RuntimeHost.h
+++ b/apps/LoopThroughWithOpenGLCompositing/RuntimeHost.h
@@ -38,6 +38,7 @@ public:
 	void SetDeckLinkOutputStatus(const std::string& modelName, bool supportsInternalKeying, bool supportsExternalKeying,
 		bool keyerInterfaceAvailable, bool externalKeyingRequested, bool externalKeyingActive, const std::string& statusMessage);
 	void SetPerformanceStats(double frameBudgetMilliseconds, double renderMilliseconds);
+	void SetAudioStatus(const AudioStatusSnapshot& status);
 	void AdvanceFrame();

 	bool BuildLayerFragmentShaderSource(const std::string& layerId, std::string& fragmentShaderSource, std::string& error);
@@ -52,6 +53,14 @@ public:
 	unsigned short GetOscPort() const { return mConfig.oscPort; }
 	unsigned GetMaxTemporalHistoryFrames() const { return mConfig.maxTemporalHistoryFrames; }
 	bool ExternalKeyingEnabled() const { return mConfig.enableExternalKeying; }
+	bool AudioEnabled() const { return mConfig.audioEnabled; }
+	bool AudioOutputEnabled() const { return mConfig.audioOutputEnabled; }
+	bool AudioScheduleEnabled() const { return mConfig.audioScheduleEnabled; }
+	bool AudioPrerollEnabled() const { return mConfig.audioPrerollEnabled; }
+	bool AudioScheduleSilence() const { return mConfig.audioScheduleSilence; }
+	bool AudioScheduleTone() const { return mConfig.audioScheduleTone; }
+	unsigned AudioChannelCount() const { return mConfig.audioChannelCount; }
+	unsigned AudioSampleRate() const { return mConfig.audioSampleRate; }
 	const std::string& GetInputVideoFormat() const { return mConfig.inputVideoFormat; }
 	const std::string& GetInputFrameRate() const { return mConfig.inputFrameRate; }
 	const std::string& GetOutputVideoFormat() const { return mConfig.outputVideoFormat; }
@@ -68,6 +77,15 @@ private:
 		bool autoReload = true;
 		unsigned maxTemporalHistoryFrames = 4;
 		bool enableExternalKeying = false;
+		bool audioEnabled = true;
+		bool audioOutputEnabled = true;
+		bool audioScheduleEnabled = true;
+		bool audioPrerollEnabled = true;
+		bool audioScheduleSilence = false;
+		bool audioScheduleTone = false;
+		unsigned audioChannelCount = kAudioChannelCount;
+		unsigned audioSampleRate = kAudioSampleRate;
+		std::string audioDelayMode = "matchVideoPreroll";
 		std::string inputVideoFormat = "1080p";
 		std::string inputFrameRate = "59.94";
 		std::string outputVideoFormat = "1080p";
@@ -148,6 +166,7 @@ private:
 	double mRenderMilliseconds;
 	double mSmoothedRenderMilliseconds;
 	DeckLinkOutputStatus mDeckLinkOutputStatus;
+	AudioStatusSnapshot mAudioStatus;
 	unsigned short mServerPort;
 	bool mAutoReloadEnabled;
 	std::chrono::steady_clock::time_point mStartTime;
--- a/apps/LoopThroughWithOpenGLCompositing/ShaderTypes.h
+++ b/apps/LoopThroughWithOpenGLCompositing/ShaderTypes.h
@@ -5,6 +5,8 @@
 #include <string>
 #include <vector>

+#include "AudioSupport.h"
+
 enum class ShaderParameterType
 {
 	Float,
@@ -95,6 +97,7 @@ struct RuntimeRenderState
 	unsigned inputHeight = 0;
 	unsigned outputWidth = 0;
 	unsigned outputHeight = 0;
+	AudioAnalysisSnapshot audioAnalysis;
 	bool isTemporal = false;
 	TemporalHistorySource temporalHistorySource = TemporalHistorySource::None;
 	unsigned requestedTemporalHistoryLength = 0;
--- a/config/runtime-host.json
+++ b/config/runtime-host.json
@@ -8,5 +8,14 @@
  "outputFrameRate": "59.94",
  "autoReload": true,
  "maxTemporalHistoryFrames": 12,
+  "audioEnabled": true,
+  "audioOutputEnabled": true,
+  "audioScheduleEnabled": true,
+  "audioPrerollEnabled": true,
+  "audioScheduleSilence": false,
+  "audioScheduleTone": false,
+  "audioChannelCount": 2,
+  "audioSampleRate": 48000,
+  "audioDelayMode": "matchVideoPreroll",
  "enableExternalKeying": true
 }
--- a/docs/AUDIO_TEARING_INVESTIGATION.md
+++ b/docs/AUDIO_TEARING_INVESTIGATION.md
@@ -0,0 +1,406 @@
+# Audio / SDI Tearing Investigation
+
+Date: 2026-05-05
+
+## Problem
+
+After adding DeckLink audio pass-through, the SDI output intermittently shows a torn/corrupted frame. The preview window does not show the artifact.
+
+Observed artifact:
+
+- Bottom portion of the SDI image can show an offset mix of current/previous frame.
+- Looks like a frame-buffer or output-transfer issue rather than shader rendering.
+- Occurs even with all shaders bypassed.
+- Main branch is known good with no tearing.
+
+Later tests also showed audio tearing/stutter when non-silent audio was scheduled.
+
+## Known Good Baseline
+
+- `main` branch has no SDI tearing.
+- Current branch with `audioEnabled: false` ran for several minutes with no visible tearing.
+
+This strongly suggests the issue is tied to DeckLink audio output/scheduling rather than the shader stack.
+
+## SDK References Checked
+
+### `InputLoopThrough`
+
+Location:
+
+`3rdParty/Blackmagic DeckLink SDK 16.0/Win/Samples/InputLoopThrough`
+
+Findings:
+
+- This is the SDK loop-through sample that keeps audio.
+- It preserves DeckLink audio packet timestamps using `GetPacketTime(..., m_frameTimescale)`.
+- It schedules audio packets with `ScheduleAudioSamples(..., packetTime, m_frameTimescale, ...)`.
+- It uses 16-channel 32-bit embedded audio by default.
+- It has separate scheduling threads for video/audio.
+- It waits for both video and audio preroll before `StartScheduledPlayback`.
+
+### `LoopThroughWithOpenGLCompositing`
+
+Location:
+
+`3rdParty/Blackmagic DeckLink SDK 16.0/Win/Samples/LoopThroughWithOpenGLCompositing`
+
+Findings:
+
+- This sample is the base for this app.
+- It ignores `IDeckLinkAudioInputPacket`.
+- It does not demonstrate audio pass-through.
+
+### `SignalGenerator`
+
+Location:
+
+`3rdParty/Blackmagic DeckLink SDK 16.0/Win/Samples/SignalGenerator`
+
+Findings:
+
+- Uses `RenderAudioSamples()` callback to top up audio when DeckLink requests samples.
+- Uses `GetBufferedAudioSampleFrameCount()` and a water level before scheduling more audio.
+
+## Tests Tried And Results
+
+### 1. Initial audio pass-through with FIFO and sample-time accumulator
+
+Implementation:
+
+- Copied incoming audio into a stereo FIFO.
+- Scheduled audio with a generated `mNextAudioSampleFrame` clock in 48 kHz timescale.
+- Matched delay to video preroll.
+
+Result:
+
+- Audio eventually worked.
+- SDI video tearing appeared.
+
+Conclusion:
+
+- Basic audio output path triggered SDI instability.
+
+### 2. Reworked audio toward SDK `InputLoopThrough` packet-timestamp model
+
+Implementation:
+
+- Preserved incoming packet time via `GetPacketTime(..., mFrameTimescale)`.
+- Queued timestamped audio packets.
+- Scheduled packets with `ScheduleAudioSamples(..., packet.streamTime, mFrameTimescale, ...)`.
+
+Result:
+
+- Tearing persisted.
+
+Conclusion:
+
+- Simply matching SDK timestamp domain did not fix the issue.
+
+### 3. Restored video callback closer to `main`
+
+Implementation:
+
+- Removed extra `glFinish()` calls.
+- Restored preview/readback ordering closer to `main`.
+- Re-enabled fast transfer path after earlier tests disabled it.
+- Removed audio texture upload from video playout callback.
+- Removed audio analysis and audio locks from video playout callback.
+- Removed DeckLink scheduling mutex around `ScheduleVideoFrame`.
+
+Result:
+
+- Tearing frequency seemed reduced at one point, but tearing persisted.
+
+Conclusion:
+
+- Extra work in the playout callback may have made timing worse, but was not the root cause.
+
+### 4. Disabled audio completely
+
+Config:
+
+```json
+"audioEnabled": false
+```
+
+Result:
+
+- Ran for several minutes with no visible tearing.
+
+Conclusion:
+
+- The tearing is tied to audio being enabled.
+
+### 5. Enabled audio input/analysis but disabled DeckLink audio output
+
+Config:
+
+```json
+"audioEnabled": true,
+"audioOutputEnabled": false
+```
+
+Result:
+
+- No tearing appeared.
+
+Conclusion:
+
+- DeckLink audio input and CPU analysis are not the trigger.
+- The problem is on the DeckLink audio output side.
+
+### 6. Enabled DeckLink audio output but disabled scheduling
+
+Config:
+
+```json
+"audioEnabled": true,
+"audioOutputEnabled": true,
+"audioScheduleEnabled": false
+```
+
+Result:
+
+- No video tearing.
+- Slight stutter appeared.
+
+Conclusion:
+
+- `EnableAudioOutput()` alone did not produce the tearing.
+- Stutter was likely from enabling an audio output stream without feeding it samples.
+
+### 7. Enabled audio scheduling but skipped audio preroll
+
+Config:
+
+```json
+"audioEnabled": true,
+"audioOutputEnabled": true,
+"audioScheduleEnabled": true,
+"audioPrerollEnabled": false
+```
+
+Result:
+
+- Video tearing returned.
+- Stutter also present.
+
+Conclusion:
+
+- `BeginAudioPreroll()` / `EndAudioPreroll()` are not required to trigger the tear.
+- `ScheduleAudioSamples()` is strongly implicated.
+
+### 8. Retained scheduled audio packet memory after `ScheduleAudioSamples`
+
+Implementation:
+
+- Kept scheduled packet buffers alive in a retain queue after scheduling.
+- Avoided passing DeckLink pointers to vectors that immediately went out of scope.
+
+Result:
+
+- Video tearing and stutter persisted.
+
+Conclusion:
+
+- Buffer lifetime after `ScheduleAudioSamples()` was not the root cause.
+
+### 9. Added audio water-level cap
+
+Implementation:
+
+- Restored SDK-style `GetBufferedAudioSampleFrameCount()` check.
+- Only scheduled more audio if DeckLink buffer was below the target water level.
+
+Result:
+
+- Stutter was reduced.
+- Video tearing persisted.
+
+Conclusion:
+
+- Overscheduling contributed to stutter/timing pressure.
+- It did not explain the tearing.
+
+### 10. Removed standalone audio scheduler thread
+
+Implementation:
+
+- Stopped starting the dedicated audio scheduler thread.
+- Audio top-up occurred from input packet arrival and `RenderAudioSamples()` callback.
+
+Result:
+
+- No meaningful change.
+
+Conclusion:
+
+- The polling thread itself was not the cause.
+
+### 11. Switched from timestamped audio output to continuous audio output
+
+Implementation:
+
+- Changed audio output to `bmdAudioOutputStreamContinuous`.
+- Scheduled audio using a monotonic 48 kHz sample clock.
+
+Result:
+
+- Video tearing and stutter persisted.
+
+Conclusion:
+
+- The issue was not specific to timestamped output mode.
+
+### 12. Rendered into the actual `completedFrame`
+
+Implementation:
+
+- Changed `PlayoutFrameCompleted()` to reuse the exact `completedFrame` passed by DeckLink rather than rotating an independent output-frame queue.
+
+Result:
+
+- No change.
+
+Conclusion:
+
+- The app was probably not overwriting a still-in-use frame from its output queue.
+
+### 13. Scheduled generated silence instead of captured audio
+
+Config:
+
+```json
+"audioScheduleSilence": true
+```
+
+Result:
+
+- Occasional stutter.
+- No video tearing.
+
+Conclusion:
+
+- Scheduling audio buffers itself can be stable if the audio data is zero.
+- Non-zero audio data appears to be important.
+
+### 14. Flattened captured audio into PCM FIFO and scheduled fixed chunks
+
+Implementation:
+
+- Captured packets were flattened into a PCM FIFO.
+- DeckLink received fixed 10 ms chunks rather than original packet boundaries.
+- Missing audio was padded with silence.
+
+Result:
+
+- Video tearing returned.
+- Audio stutter/tearing returned.
+
+Conclusion:
+
+- Packet boundaries/timestamps were not the whole cause.
+- Non-zero captured audio data still triggered instability.
+
+### 15. Scheduled generated 440 Hz tone
+
+Config:
+
+```json
+"audioScheduleTone": true
+```
+
+Result:
+
+- Video tearing occurred.
+- Tone/audio also tore.
+
+Conclusion:
+
+- The issue is not specific to captured input data.
+- Non-zero scheduled audio, even generated tone, triggers the problem.
+
+### 16. Changed DeckLink output to 16 embedded audio channels
+
+Implementation:
+
+- Enabled DeckLink audio output with 16 channels instead of 2.
+- Mapped stereo to channels 1/2.
+- Filled channels 3-16 with silence.
+
+Result:
+
+- Video tearing and audio tearing still occurred.
+
+Conclusion:
+
+- The issue is not simply caused by 2-channel embedded audio output.
+
+### 17. Used DeckLink-owned output video frames with audio enabled
+
+Implementation:
+
+- When audio output is enabled:
+  - disabled fast transfer path
+  - created output frames with `CreateVideoFrame()`
+  - avoided `CreateVideoFrameWithBuffer()` and the custom pinned playout allocator
+
+Result:
+
+- Video tearing and audio tearing still occurred.
+
+Conclusion:
+
+- The custom pinned output video buffers are likely not the root cause.
+
+## Current Strong Conclusions
+
+- Shader stack is not the cause.
+- Preview/render output is not showing the issue, so the artifact is SDI/output-side.
+- DeckLink audio input is not the cause.
+- DeckLink audio output enabled but unscheduled does not cause tearing.
+- `ScheduleAudioSamples()` with zero/silent buffers does not cause tearing.
+- `ScheduleAudioSamples()` with non-zero audio causes both video tearing and audio tearing.
+- The problem persists across:
+  - timestamped audio output
+  - continuous audio output
+  - captured audio
+  - generated tone
+  - 2-channel output
+  - 16-channel embedded output
+  - app-owned/pinned output video buffers
+  - DeckLink-owned output video frames
+
+## Current Hypothesis
+
+The issue appears to be a DeckLink output interaction where non-zero embedded audio samples disturb SDI video/audio output in this app’s scheduling model.
+
+Since silence is stable but tone is not, the next likely areas to investigate are:
+
+- Audio sample format/range/endian expectations.
+- Whether DeckLink expects 32-bit audio samples to be in a different effective range than we are providing.
+- Whether the scheduled audio buffer layout for the selected hardware/output mode differs from our assumptions.
+- Whether the selected output mode/keyer/SDI configuration has constraints when non-zero embedded audio is present.
+- Whether the SDK sample behaves correctly on the same hardware with a generated tone and same video mode.
+
+## Suggested Next Tests
+
+1. Schedule very low amplitude non-zero audio, e.g. constant `1`, then `256`, then a very quiet sine.
+2. Try 16-bit audio output instead of 32-bit if supported.
+3. Try `bmdAudioOutputStreamContinuousDontResample`.
+4. Disable external keying and test with non-zero audio.
+5. Build/run the SDK `SignalGenerator` or `InputLoopThrough` sample on the same DeckLink device, video mode, and SDI output path with non-zero embedded audio.
+6. Add instrumentation for DeckLink status/errors around scheduled video/audio completion.
+7. Confirm Desktop Video setup panel audio/SDI settings for the selected output.
+
+## Current Config At Time Of Note
+
+```json
+"audioEnabled": true,
+"audioOutputEnabled": true,
+"audioScheduleEnabled": true,
+"audioPrerollEnabled": true,
+"audioScheduleSilence": false,
+"audioScheduleTone": false
+```
--- a/runtime/templates/shader_wrapper.slang.in
+++ b/runtime/templates/shader_wrapper.slang.in
@@ -16,6 +16,11 @@ struct ShaderContext
 	float bypass;
 	int sourceHistoryLength;
 	int temporalHistoryLength;
+	float2 audioRms;
+	float2 audioPeak;
+	float audioMonoRms;
+	float audioMonoPeak;
+	float4 audioBands;
 };

 cbuffer GlobalParams
@@ -28,15 +33,31 @@ cbuffer GlobalParams
 	float gBypass;
 	int gSourceHistoryLength;
 	int gTemporalHistoryLength;
+	float2 gAudioRms;
+	float2 gAudioPeak;
+	float gAudioMonoRms;
+	float gAudioMonoPeak;
+	float4 gAudioBands;
 {{PARAMETER_UNIFORMS}}};

 Sampler2D<float4> gVideoInput;
+Sampler2D<float4> gAudioData;
 {{SOURCE_HISTORY_SAMPLERS}}{{TEMPORAL_HISTORY_SAMPLERS}}{{TEXTURE_SAMPLERS}}
 float4 sampleVideo(float2 tc)
 {
 	return gVideoInput.Sample(tc);
 }

+float4 sampleAudioWaveform(float x)
+{
+	return gAudioData.Sample(float2(saturate(x), 0.25));
+}
+
+float4 sampleAudioSpectrum(float x)
+{
+	return gAudioData.Sample(float2(saturate(x), 0.75));
+}
+
 float4 sampleSourceHistory(int framesAgo, float2 tc)
 {
 	if (gSourceHistoryLength <= 0)
@@ -83,6 +104,11 @@ float4 fragmentMain(FragmentInput input) : SV_Target
 	context.bypass = gBypass;
 	context.sourceHistoryLength = gSourceHistoryLength;
 	context.temporalHistoryLength = gTemporalHistoryLength;
+	context.audioRms = gAudioRms;
+	context.audioPeak = gAudioPeak;
+	context.audioMonoRms = gAudioMonoRms;
+	context.audioMonoPeak = gAudioMonoPeak;
+	context.audioBands = gAudioBands;
 	float4 effectedColor = {{ENTRY_POINT_CALL}};
 	float mixValue = clamp(gBypass > 0.5 ? 0.0 : gMixAmount, 0.0, 1.0);
 	return lerp(context.sourceColor, effectedColor, mixValue);
--- a/shaders/audio-vu-meter/shader.json
+++ b/shaders/audio-vu-meter/shader.json
@@ -0,0 +1,76 @@
+{
+  "id": "audio-vu-meter",
+  "name": "Audio VU Meter",
+  "description": "Draws stereo audio level meters from the runtime audio analysis data.",
+  "category": "Utility",
+  "entryPoint": "shadeVideo",
+  "parameters": [
+    {
+      "id": "meterPosition",
+      "label": "Position",
+      "type": "vec2",
+      "default": [0.08, 0.82],
+      "min": [0.0, 0.0],
+      "max": [1.0, 1.0],
+      "step": [0.01, 0.01]
+    },
+    {
+      "id": "meterScale",
+      "label": "Scale",
+      "type": "float",
+      "default": 0.35,
+      "min": 0.1,
+      "max": 1.0,
+      "step": 0.01
+    },
+    {
+      "id": "meterOpacity",
+      "label": "Opacity",
+      "type": "float",
+      "default": 0.9,
+      "min": 0.0,
+      "max": 1.0,
+      "step": 0.01
+    },
+    {
+      "id": "noiseGate",
+      "label": "Noise Gate",
+      "type": "float",
+      "default": 0.03,
+      "min": 0.0,
+      "max": 0.5,
+      "step": 0.01
+    },
+    {
+      "id": "meterColor",
+      "label": "Meter Color",
+      "type": "color",
+      "default": [0.2, 1.0, 0.55, 1.0]
+    },
+    {
+      "id": "peakColor",
+      "label": "Peak Color",
+      "type": "color",
+      "default": [1.0, 0.85, 0.2, 1.0]
+    },
+    {
+      "id": "backgroundOpacity",
+      "label": "Background",
+      "type": "float",
+      "default": 0.45,
+      "min": 0.0,
+      "max": 1.0,
+      "step": 0.01
+    },
+    {
+      "id": "orientation",
+      "label": "Orientation",
+      "type": "enum",
+      "default": "horizontal",
+      "options": [
+        { "value": "horizontal", "label": "Horizontal" },
+        { "value": "vertical", "label": "Vertical" }
+      ]
+    }
+  ]
+}
--- a/shaders/audio-vu-meter/shader.slang
+++ b/shaders/audio-vu-meter/shader.slang
@@ -0,0 +1,59 @@
+float rectMask(float2 uv, float2 minUv, float2 maxUv)
+{
+	float2 insideMin = step(minUv, uv);
+	float2 insideMax = step(uv, maxUv);
+	return insideMin.x * insideMin.y * insideMax.x * insideMax.y;
+}
+
+float denoiseLevel(float value)
+{
+	float gate = saturate(noiseGate);
+	float clean = saturate((value - gate) / max(1.0 - gate, 0.001));
+	return smoothstep(0.0, 1.0, clean);
+}
+
+float4 shadeVideo(ShaderContext context)
+{
+	float4 color = context.sourceColor;
+	float2 size = orientation == 0 ? float2(meterScale, meterScale * 0.18) : float2(meterScale * 0.18, meterScale);
+	float2 minUv = clamp(meterPosition, 0.0, 1.0 - size);
+	float2 local = (context.uv - minUv) / max(size, float2(0.001));
+	float inside = rectMask(local, float2(0.0), float2(1.0));
+	if (inside <= 0.0)
+		return color;
+
+	float3 bg = lerp(color.rgb, float3(0.0), saturate(backgroundOpacity));
+	float leftLevel = denoiseLevel(context.audioRms.x * 2.4);
+	float rightLevel = denoiseLevel(context.audioRms.y * 2.4);
+	float leftPeak = denoiseLevel(context.audioPeak.x);
+	float rightPeak = denoiseLevel(context.audioPeak.y);
+
+	float bar = 0.0;
+	float peak = 0.0;
+	if (orientation == 0)
+	{
+		float leftRow = rectMask(local, float2(0.04, 0.58), float2(0.96, 0.86));
+		float rightRow = rectMask(local, float2(0.04, 0.14), float2(0.96, 0.42));
+		float leftFill = rectMask(local, float2(0.04, 0.58), float2(0.04 + 0.92 * leftLevel, 0.86));
+		float rightFill = rectMask(local, float2(0.04, 0.14), float2(0.04 + 0.92 * rightLevel, 0.42));
+		float leftPeakLine = rectMask(local, float2(0.04 + 0.92 * leftPeak - 0.006, 0.55), float2(0.04 + 0.92 * leftPeak + 0.006, 0.89));
+		float rightPeakLine = rectMask(local, float2(0.04 + 0.92 * rightPeak - 0.006, 0.11), float2(0.04 + 0.92 * rightPeak + 0.006, 0.45));
+		bar = max(leftFill, rightFill);
+		peak = max(leftPeakLine * leftRow, rightPeakLine * rightRow);
+	}
+	else
+	{
+		float leftColumn = rectMask(local, float2(0.14, 0.04), float2(0.42, 0.96));
+		float rightColumn = rectMask(local, float2(0.58, 0.04), float2(0.86, 0.96));
+		float leftFill = rectMask(local, float2(0.14, 0.04), float2(0.42, 0.04 + 0.92 * leftLevel));
+		float rightFill = rectMask(local, float2(0.58, 0.04), float2(0.86, 0.04 + 0.92 * rightLevel));
+		float leftPeakLine = rectMask(local, float2(0.11, 0.04 + 0.92 * leftPeak - 0.006), float2(0.45, 0.04 + 0.92 * leftPeak + 0.006));
+		float rightPeakLine = rectMask(local, float2(0.55, 0.04 + 0.92 * rightPeak - 0.006), float2(0.89, 0.04 + 0.92 * rightPeak + 0.006));
+		bar = max(leftFill * leftColumn, rightFill * rightColumn);
+		peak = max(leftPeakLine, rightPeakLine);
+	}
+
+	float3 metered = lerp(bg, meterColor.rgb, bar * saturate(meterOpacity) * meterColor.a);
+	metered = lerp(metered, peakColor.rgb, peak * saturate(meterOpacity) * peakColor.a);
+	return float4(metered, color.a);
+}
--- a/tests/AudioSupportTests.cpp
+++ b/tests/AudioSupportTests.cpp
@@ -0,0 +1,115 @@
+#include "AudioSupport.h"
+
+#include <algorithm>
+#include <cmath>
+#include <cstdint>
+#include <iostream>
+#include <vector>
+
+namespace
+{
+int gFailures = 0;
+
+void Expect(bool condition, const char* message)
+{
+	if (condition)
+		return;
+
+	std::cerr << "FAIL: " << message << "\n";
+	++gFailures;
+}
+
+int32_t ToSample(float value)
+{
+	const double clamped = std::max(-1.0, std::min(1.0, static_cast<double>(value)));
+	return static_cast<int32_t>(clamped * 2147483647.0);
+}
+
+void TestFrameSampleCounts()
+{
+	Expect(AudioSamplesForVideoFrame(0, 1, 50) == 960, "50 fps first frame has 960 audio samples");
+	Expect(AudioSamplesForVideoFrame(0, 1, 60) == 800, "60 fps first frame has 800 audio samples");
+
+	uint64_t total = 0;
+	for (uint64_t frame = 0; frame < 600; ++frame)
+		total += AudioSamplesForVideoFrame(frame, 1001, 60000);
+	Expect(total == AudioSampleTimeForVideoFrame(600, 1001, 60000), "59.94 fps sample counts do not drift");
+}
+
+void TestDelayBuffer()
+{
+	AudioDelayBuffer buffer;
+	buffer.Reset(4);
+	std::vector<int32_t> input = {
+		11, 12,
+		21, 22,
+		31, 32,
+		41, 42
+	};
+	buffer.PushInterleaved(input.data(), 4);
+
+	bool underrun = false;
+	AudioFrameBlock first = buffer.Pop(4, underrun);
+	Expect(!underrun, "delay-buffer initial silence does not underrun");
+	Expect(first.frameCount() == 4, "delay-buffer returns requested frame count");
+	Expect(first.interleavedSamples[0] == 0 && first.interleavedSamples[7] == 0, "delay-buffer emits initial silence");
+
+	AudioFrameBlock second = buffer.Pop(4, underrun);
+	Expect(!underrun, "delay-buffer emits delayed input without underrun");
+	Expect(second.interleavedSamples == input, "delay-buffer preserves delayed interleaved samples");
+
+	AudioFrameBlock third = buffer.Pop(2, underrun);
+	Expect(underrun, "delay-buffer reports underrun");
+	Expect(third.interleavedSamples[0] == 0 && third.interleavedSamples[3] == 0, "delay-buffer underrun fills silence");
+}
+
+void TestAnalyzerSilence()
+{
+	AudioAnalyzer analyzer;
+	AudioFrameBlock block;
+	block.interleavedSamples.resize(512 * kAudioChannelCount, 0);
+
+	AudioAnalysisSnapshot analysis = analyzer.Analyze(block);
+	Expect(analysis.rms[0] == 0.0f && analysis.rms[1] == 0.0f, "silence rms is zero");
+	Expect(analysis.peak[0] == 0.0f && analysis.peak[1] == 0.0f, "silence peak is zero");
+	Expect(analysis.bands[0] == 0.0f && analysis.bands[3] == 0.0f, "silence bands are zero");
+}
+
+void TestAnalyzerSineAndStereo()
+{
+	AudioAnalyzer analyzer;
+	AudioFrameBlock block;
+	block.interleavedSamples.resize(1024 * kAudioChannelCount, 0);
+
+	for (std::size_t frame = 0; frame < 1024; ++frame)
+	{
+		const float phase = static_cast<float>(frame) * 2.0f * 3.14159265f * 300.0f / static_cast<float>(kAudioSampleRate);
+		block.interleavedSamples[frame * 2] = ToSample(std::sin(phase) * 0.8f);
+		block.interleavedSamples[frame * 2 + 1] = ToSample(0.25f);
+	}
+
+	AudioAnalysisSnapshot analysis = analyzer.Analyze(block);
+	Expect(analysis.peak[0] > 0.75f && analysis.peak[0] <= 0.81f, "left sine peak is detected");
+	Expect(analysis.rms[0] > 0.45f && analysis.rms[0] < 0.65f, "left sine rms is detected");
+	Expect(analysis.peak[1] > 0.24f && analysis.peak[1] < 0.26f, "right constant peak remains independent");
+	Expect(analysis.rms[1] > 0.24f && analysis.rms[1] < 0.26f, "right constant rms remains independent");
+	Expect(analysis.bands[1] >= analysis.bands[0], "300 Hz sine activates lower-mid band");
+}
+}
+
+int main()
+{
+	TestFrameSampleCounts();
+	TestDelayBuffer();
+	TestAnalyzerSilence();
+	TestAnalyzerSineAndStereo();
+
+	if (gFailures != 0)
+	{
+		std::cerr << gFailures << " AudioSupport test failure(s).\n";
+		return 1;
+	}
+
+	std::cout << "AudioSupport tests passed.\n";
+	return 0;
+}
--- a/ui/src/App.jsx
+++ b/ui/src/App.jsx
@@ -19,6 +19,7 @@ function App() {
  const performance = appState?.performance ?? {};
  const runtime = appState?.runtime ?? {};
  const video = appState?.video ?? {};
+  const audio = appState?.audio ?? {};
  const app = appState?.app ?? {};
  const stackPresets = appState?.stackPresets ?? [];

@@ -67,7 +68,7 @@ function App() {
      </header>

      <section className="dashboard-grid">
-        <StatusPanels app={app} performance={performance} runtime={runtime} video={video} />
+        <StatusPanels app={app} audio={audio} performance={performance} runtime={runtime} video={video} />
        <StackPresetToolbar
          presetName={presetName}
          selectedPresetName={selectedPresetName}
--- a/ui/src/components/StatusPanels.jsx
+++ b/ui/src/components/StatusPanels.jsx
@@ -4,7 +4,7 @@ function formatNumber(value, digits = 3) {
  return Number(value ?? 0).toFixed(digits);
 }

-export function StatusPanels({ app, performance, runtime, video }) {
+export function StatusPanels({ app, audio, performance, runtime, video }) {
  return (
    <>
      <div className="panel panel--runtime">
@@ -36,6 +36,21 @@ export function StatusPanels({ app, performance, runtime, video }) {
        />
      </div>

+      <div className="panel panel--audio">
+        <h2>Audio</h2>
+        <KvList
+          values={[
+            ["Enabled", audio.enabled ? "On" : "Off"],
+            ["Sample Rate", `${app.audioSampleRate || 0} Hz`],
+            ["Channels", `${app.audioChannelCount || 0}`],
+            ["Buffered", `${audio.bufferedSampleFrames || 0} samples`],
+            ["Underruns", `${audio.underrunCount || 0}`],
+            ["RMS L/R", `${formatNumber(audio.rms?.[0], 3)} / ${formatNumber(audio.rms?.[1], 3)}`],
+            ["Peak L/R", `${formatNumber(audio.peak?.[0], 3)} / ${formatNumber(audio.peak?.[1], 3)}`],
+          ]}
+        />
+      </div>
+
      <div className="panel panel--compiler">
        <h2>Compiler</h2>
        <pre>{runtime.compileMessage || "No compiler output."}</pre>
Author	SHA1	Message	Date
Aiden	9a8748687a	Audio experiments	2026-05-05 12:18:42 +10:00
Aiden	f836c53d10	Initial audio support	2026-05-04 14:32:29 +10:00