audio: WASAPI exclusive resampling, buffer size increase options (#727)

## Link to GitHub Issue or related Pull Request, if one exists
n/a

## Description of change

Resampler:
Implement resampler for exclusive mode streams, as we are seeing more
and more devices - not just laptops but onboard audio devices - that
only support 48khz and not 44.1khz. Should work with volume boost (gain
calculated inside resample) and also downmixer (hands off intermediate
scratch buffers).

Buffer size increase:
By default many of these games request a tiny buffer when in shared mode
(TDJ uses 3ms). On some audio setup this results in crackling due to
underflow. Add an option to forcibly increase the buffer size.

## Testing
With resampler set to 48kHz and buffer set to 20ms I can reliably boot
and play IIDX on my display port monitor's speakers; previously this
wasn't possible. IIDX is event-driven.

Tested SDVX7 as well at 48kHz, which opens timer-driven streams.

src/spice2x/CMakeLists.txt

@@ -505,6 +505,7 @@ set(SOURCE_FILES ${SOURCE_FILES}
         hooks/audio/backends/wasapi/audio_client.cpp
         hooks/audio/backends/wasapi/audio_render_client.cpp
         hooks/audio/backends/wasapi/downmix.cpp
+        hooks/audio/backends/wasapi/resample.cpp
         hooks/audio/backends/wasapi/dummy_audio_client.cpp
         hooks/audio/backends/wasapi/dummy_audio_clock.cpp
         hooks/audio/backends/wasapi/dummy_audio_render_client.cpp

src/spice2x/hooks/audio/audio.cpp

@@ -41,6 +41,8 @@ namespace hooks::audio {
     bool VOLUME_HOOK_ENABLED = true;
     std::optional<DownmixAlgorithm> DOWNMIX_ALGORITHM = std::nullopt;
     float VOLUME_BOOST = 1.0f;
+    std::optional<uint32_t> RESAMPLE_RATE = std::nullopt;
+    std::optional<uint32_t> EXCLUSIVE_BUFFER_MS = std::nullopt;
     bool USE_DUMMY = false;
     WAVEFORMATEXTENSIBLE FORMAT {};
     std::optional<Backend> BACKEND = std::nullopt;

src/spice2x/hooks/audio/audio.h

@@ -1,5 +1,6 @@
 #pragma once
 
+#include <cstdint>
 #include <optional>
 #include <string>
 
@@ -29,6 +30,13 @@ namespace hooks::audio {
     extern bool VOLUME_HOOK_ENABLED;
     extern std::optional<DownmixAlgorithm> DOWNMIX_ALGORITHM;
     extern float VOLUME_BOOST;
+
+    // target sample rate the hooked output is resampled to, if set
+    extern std::optional<uint32_t> RESAMPLE_RATE;
+
+    // minimum WASAPI exclusive buffer duration (milliseconds), if set. enlarges the device buffer
+    // to avoid underrun crackle on endpoints that cannot service a tiny buffer in time.
+    extern std::optional<uint32_t> EXCLUSIVE_BUFFER_MS;
     extern bool USE_DUMMY;
     extern WAVEFORMATEXTENSIBLE FORMAT;
     extern std::optional<Backend> BACKEND;

src/spice2x/hooks/audio/backends/wasapi/audio_client.cpp

@@ -181,13 +181,22 @@ HRESULT STDMETHODCALLTYPE WrappedIAudioClient::Initialize(
         games::gitadora::fix_audio_channel_mask(const_cast<WAVEFORMATEX *>(pFormat));
     }
 
+    // when resampling, open the real device at the target rate while the game keeps writing its
+    // native-rate audio into the scratch buffer. this runs on whatever device_format is now: the
+    // game's native format, or the stereo format produced above when downmix is also active, so
+    // the two stages chain as multi-channel -> stereo -> resampled stereo.
+    WAVEFORMATEXTENSIBLE resample_storage = {};
+    if (auto target_rate = hooks::audio::Resampler::resolve(device_format)) {
+        const uint32_t src_rate = device_format->nSamplesPerSec;
+        this->resample.setup(device_format, &resample_storage, *target_rate);
+        device_format = reinterpret_cast<const WAVEFORMATEX *>(&resample_storage);
+        log_info("audio::wasapi", "resample enabled: {} Hz -> {} Hz{}",
+                src_rate, *target_rate, this->downmix.enabled ? " (after downmix)" : "");
+    }
+
     // verbose output
     log_info("audio::wasapi", "IAudioClient::Initialize hook hit");
-    log_info("audio::wasapi", "... ShareMode         : {}", share_mode_str(ShareMode));
+    print_format(ShareMode, StreamFlags, hnsBufferDuration, hnsPeriodicity, device_format);
-    log_info("audio::wasapi", "... StreamFlags       : {}", stream_flags_str(StreamFlags));
-    log_info("audio::wasapi", "... hnsBufferDuration : {}", hnsBufferDuration);
-    log_info("audio::wasapi", "... hnsPeriodicity    : {}", hnsPeriodicity);
-    print_format(device_format);
 
     if (this->backend) {
         SAFE_CALL("AudioBackend", "on_initialize", this->backend->on_initialize(
@@ -199,22 +208,48 @@ HRESULT STDMETHODCALLTYPE WrappedIAudioClient::Initialize(
                 AudioSessionGuid));
 
         log_info("audio::wasapi", "AudioBackend::on_initialize call finished");
-        log_info("audio::wasapi", "... ShareMode         : {}", share_mode_str(ShareMode));
+        print_format(ShareMode, StreamFlags, hnsBufferDuration, hnsPeriodicity, pFormat);
-        log_info("audio::wasapi", "... StreamFlags       : {}", stream_flags_str(StreamFlags));
-        log_info("audio::wasapi", "... hnsBufferDuration : {}", hnsBufferDuration);
-        log_info("audio::wasapi", "... hnsPeriodicity    : {}", hnsPeriodicity);
-        print_format(pFormat);
     }
 
     // check for exclusive mode
     if (ShareMode == AUDCLNT_SHAREMODE_EXCLUSIVE) {
         this->exclusive_mode = true;
         this->frame_size = device_format->nChannels * (device_format->wBitsPerSample / 8);
+
+        // optionally enlarge the exclusive buffer. games request a very small buffer (e.g. 3 ms)
+        // which some endpoints (notably NVIDIA HDMI/DP display audio) cannot service in time,
+        // underrunning mid-period and crackling. a larger buffer gives the device slack. exclusive
+        // mode requires periodicity == buffer_duration, so raise both together; the initialize
+        // paths below handle any required buffer-size realignment.
+        if (hooks::audio::EXCLUSIVE_BUFFER_MS.has_value()) {
+            const REFERENCE_TIME min_duration =
+                    (REFERENCE_TIME) hooks::audio::EXCLUSIVE_BUFFER_MS.value() * 10000;
+            if (hnsBufferDuration < min_duration) {
+                log_info("audio::wasapi",
+                        "raising exclusive buffer from {} hns to {} hns ({} ms)",
+                        hnsBufferDuration, min_duration, hooks::audio::EXCLUSIVE_BUFFER_MS.value());
+                hnsBufferDuration = min_duration;
+                if (hnsPeriodicity != 0) {
+                    hnsPeriodicity = min_duration;
+                }
+            }
+        }
     }
 
-    // call next
+    // call next. the resampler owns the device interaction whenever it is active (including when
+    // chained after the downmix), otherwise the downmix does, otherwise the device is opened
+    // directly.
     HRESULT ret;
-    if (this->downmix.enabled) {
+    if (this->resample.enabled) {
+        ret = this->resample.initialize(
+                pReal,
+                ShareMode,
+                StreamFlags,
+                hnsBufferDuration,
+                hnsPeriodicity,
+                device_format,
+                AudioSessionGuid);
+    } else if (this->downmix.enabled) {
         ret = this->downmix.initialize(
                 pReal,
                 ShareMode,
@@ -224,7 +259,9 @@ HRESULT STDMETHODCALLTYPE WrappedIAudioClient::Initialize(
                 device_format,
                 AudioSessionGuid);
     } else {
-        ret = pReal->Initialize(
+        ret = initialize_with_alignment_retry(
+                pReal,
+                "audio::wasapi",
                 ShareMode,
                 StreamFlags,
                 hnsBufferDuration,
@@ -263,7 +300,15 @@ HRESULT STDMETHODCALLTYPE WrappedIAudioClient::GetBufferSize(UINT32 *pNumBufferF
         }
     }
 
-    CHECK_RESULT(pReal->GetBufferSize(pNumBufferFrames));
+    HRESULT ret = pReal->GetBufferSize(pNumBufferFrames);
+
+    // report the buffer size at the game's native rate; the real device buffer is at the
+    // resampled rate, so translate it back so the game paces its writes correctly.
+    if (SUCCEEDED(ret) && this->resample.enabled && pNumBufferFrames) {
+        *pNumBufferFrames = this->resample.frames_device_to_game(*pNumBufferFrames);
+    }
+
+    CHECK_RESULT(ret);
 }
 HRESULT STDMETHODCALLTYPE WrappedIAudioClient::GetStreamLatency(REFERENCE_TIME *phnsLatency) {
     static std::once_flag printed;
@@ -305,7 +350,15 @@ HRESULT STDMETHODCALLTYPE WrappedIAudioClient::GetCurrentPadding(UINT32 *pNumPad
         }
     }
 
-    CHECK_RESULT(pReal->GetCurrentPadding(pNumPaddingFrames));
+    HRESULT ret = pReal->GetCurrentPadding(pNumPaddingFrames);
+
+    // the device buffer is at the resampled rate; report padding at the game's native rate so the
+    // game's free-space calculation stays paced correctly.
+    if (SUCCEEDED(ret) && this->resample.enabled && pNumPaddingFrames) {
+        *pNumPaddingFrames = this->resample.padding_device_to_game(*pNumPaddingFrames);
+    }
+
+    CHECK_RESULT(ret);
 }
 HRESULT STDMETHODCALLTYPE WrappedIAudioClient::IsFormatSupported(
     AUDCLNT_SHAREMODE ShareMode,
@@ -323,16 +376,42 @@ HRESULT STDMETHODCALLTYPE WrappedIAudioClient::IsFormatSupported(
         fix_rec_format(const_cast<WAVEFORMATEX *>(pFormat));
     }
 
+    // log the format the game is asking about
+    log_info("audio::wasapi", "IAudioClient::IsFormatSupported hook hit");
+    print_format(ShareMode, pFormat);
+
     // when downmixing, the real device is opened as stereo, so check whether the equivalent
-    // stereo format is supported instead of the multi-channel one.
+    // stereo format is supported instead of the multi-channel one. when resampling is also active
+    // it chains onto that stereo format, so check the resampled stereo format.
     if (resolve_downmix(pFormat)) {
         WAVEFORMATEXTENSIBLE stereo_storage = {};
         hooks::audio::Downmix::make_stereo_format(pFormat, &stereo_storage);
-        const auto stereo_format = reinterpret_cast<const WAVEFORMATEX *>(&stereo_storage);
+        const WAVEFORMATEX *check_format = reinterpret_cast<const WAVEFORMATEX *>(&stereo_storage);
 
-        CHECK_RESULT(pReal->IsFormatSupported(ShareMode, stereo_format, ppClosestMatch));
+        WAVEFORMATEXTENSIBLE resample_storage = {};
+        if (auto target_rate = hooks::audio::Resampler::resolve(check_format)) {
+            hooks::audio::Resampler::make_device_format(check_format, &resample_storage, *target_rate);
+            check_format = reinterpret_cast<const WAVEFORMATEX *>(&resample_storage);
+        }
+
+        log_info("audio::wasapi", "... checking device format instead (after downmix/resample):");
+        print_format(check_format);
+
+        CHECK_RESULT(pReal->IsFormatSupported(ShareMode, check_format, ppClosestMatch));
     } else if (games::gitadora::is_arena_model()) {
         games::gitadora::fix_audio_channel_mask(const_cast<WAVEFORMATEX *>(pFormat));
+    } else if (auto target_rate = hooks::audio::Resampler::resolve(pFormat)) {
+
+        // when resampling, the real device is opened at the target rate, so check whether the
+        // equivalent format at that rate is supported instead of the game's native rate.
+        WAVEFORMATEXTENSIBLE resample_storage = {};
+        hooks::audio::Resampler::make_device_format(pFormat, &resample_storage, *target_rate);
+        const auto resample_format = reinterpret_cast<const WAVEFORMATEX *>(&resample_storage);
+
+        log_info("audio::wasapi", "... checking device format instead (after resample):");
+        print_format(resample_format);
+
+        CHECK_RESULT(pReal->IsFormatSupported(ShareMode, resample_format, ppClosestMatch));
     }
 
     if (this->backend) {

src/spice2x/hooks/audio/backends/wasapi/audio_client.h

@@ -9,9 +9,9 @@
 #include "util/logging.h"
 
 #include "downmix.h"
+#include "resample.h"
 
 #include "audio_render_client.h"
-
 // {1FBC8530-AF3E-4128-B418-115DE72F76B6}
 static const GUID IID_WrappedIAudioClient = {
     0x1fbc8530, 0xaf3e, 0x4128, { 0xb4, 0x18, 0x11, 0x5d, 0xe7, 0x2f, 0x76, 0xb6 }
@@ -103,4 +103,9 @@ struct WrappedIAudioClient : IAudioClient3 {
     // surround -> stereo downmix. the real device is opened as stereo while the game keeps
     // writing multi-channel audio into a scratch buffer that we downmix in the render client.
     hooks::audio::Downmix downmix;
+
+    // native-rate -> target-rate sample-rate conversion. the real device is opened at the target
+    // rate while the game keeps writing its native-rate audio into a scratch buffer that we
+    // resample in the render client.
+    hooks::audio::Resampler resample;
 };

src/spice2x/hooks/audio/backends/wasapi/audio_render_client.cpp

@@ -7,6 +7,7 @@
 
 #include "audio_client.h"
 #include "hooks/audio/audio.h"
+#include "util.h"
 #include "wasapi_private.h"
 
 const char CLASS_NAME[] = "WrappedIAudioRenderClient";
@@ -17,9 +18,7 @@ static void apply_gain(BYTE *buffer, UINT32 frames, const WAVEFORMATEXTENSIBLE &
     const WAVEFORMATEX &f = fmt.Format;
     const size_t samples = (size_t) frames * f.nChannels;
 
-    // KSDATAFORMAT_SUBTYPE_IEEE_FLOAT has Data1 == 3, _PCM has Data1 == 1
+    bool is_float = is_ieee_float(&f);
-    bool is_float = f.wFormatTag == WAVE_FORMAT_IEEE_FLOAT
-        || (f.wFormatTag == WAVE_FORMAT_EXTENSIBLE && fmt.SubFormat.Data1 == 0x00000003);
 
     if (is_float && f.wBitsPerSample == 32) {
         auto p = reinterpret_cast<float *>(buffer);
@@ -113,10 +112,23 @@ HRESULT STDMETHODCALLTYPE WrappedIAudioRenderClient::GetBuffer(UINT32 NumFramesR
         return S_OK;
     }
 
+    // downmix + resample chained: the game writes its multi-channel native-rate audio into the
+    // downmix scratch, which is downmixed to stereo and then resampled on release. size the
+    // resampler's (stereo) input scratch now and hand the game the multi-channel downmix scratch.
+    if (this->client->downmix.enabled && this->client->resample.enabled) {
+        BYTE *resample_scratch = nullptr;
+        this->client->resample.get_buffer(NumFramesRequested, &resample_scratch);
+        CHECK_RESULT(this->client->downmix.get_scratch(NumFramesRequested, ppData));
+
     // surround downmix: reserve the real (stereo) device buffer, but hand the game a
     // multi-channel scratch buffer that we downmix on release
-    if (this->client->downmix.enabled) {
+    } else if (this->client->downmix.enabled) {
         CHECK_RESULT(this->client->downmix.get_buffer(pReal, NumFramesRequested, ppData));
+
+    // resample: hand the game a native-rate scratch buffer that we convert on release. the real
+    // device buffer is acquired in ReleaseBuffer once the converted frame count is known.
+    } else if (this->client->resample.enabled) {
+        CHECK_RESULT(this->client->resample.get_buffer(NumFramesRequested, ppData));
     }
 
     // call original
@@ -143,6 +155,34 @@ HRESULT STDMETHODCALLTYPE WrappedIAudioRenderClient::ReleaseBuffer(UINT32 NumFra
         return S_OK;
     }
 
+    // downmix + resample chained: downmix the game's multi-channel scratch into the resampler's
+    // stereo input scratch, then let the resampler convert and push it to the device. a silent
+    // buffer skips the downmix and feeds silence straight through.
+    if (this->client->downmix.enabled && this->client->resample.enabled) {
+        if ((dwFlags & AUDCLNT_BUFFERFLAGS_SILENT) == 0) {
+            this->client->downmix.downmix_into(
+                    this->client->resample.input_data(), NumFramesWritten);
+        }
+        return this->client->resample.flush(
+                pReal,
+                this->client->pReal,
+                NumFramesWritten,
+                dwFlags,
+                hooks::audio::VOLUME_BOOST);
+    }
+
+    // resample: convert the game's native-rate scratch and push as many output frames as the
+    // device has room for, applying the volume boost to the converted output. handles acquiring
+    // and releasing the real device buffer itself.
+    if (this->client->resample.enabled) {
+        return this->client->resample.flush(
+                pReal,
+                this->client->pReal,
+                NumFramesWritten,
+                dwFlags,
+                hooks::audio::VOLUME_BOOST);
+    }
+
     // resolve the real device buffer for whichever path produced the audio
     BYTE *device_buffer;
     if (this->client->downmix.enabled) {

src/spice2x/hooks/audio/backends/wasapi/downmix.cpp

@@ -11,6 +11,8 @@
 
 #include "util/logging.h"
 
+#include "util.h"
+
 namespace hooks::audio {
 
     namespace {
@@ -45,69 +47,6 @@ namespace hooks::audio {
         }
     }
 
-    // read one sample at `p` as a normalized float in [-1, 1]
-    static inline float read_sample(const BYTE *p, int bytes, bool is_float) {
-        if (is_float) {
-            float v;
-            memcpy(&v, p, sizeof(float));
-            return v;
-        }
-        switch (bytes) {
-            case 2: {
-                int16_t v;
-                memcpy(&v, p, sizeof(v));
-                return v * (1.0f / 32768.0f);
-            }
-            case 3: {
-                int32_t v = p[0] | (p[1] << 8) | (p[2] << 16);
-                if (v & 0x800000) {
-                    v |= ~0xFFFFFF; // sign extend
-                }
-                return v * (1.0f / 8388608.0f);
-            }
-            case 4: {
-                int32_t v;
-                memcpy(&v, p, sizeof(v));
-                return (float) (v * (1.0 / 2147483648.0));
-            }
-            default:
-                return 0.0f;
-        }
-    }
-
-    // write the normalized float `value` to the sample at `p`, clamping to the format's range
-    static inline void write_sample(BYTE *p, int bytes, bool is_float, float value) {
-        if (is_float) {
-            float v = std::clamp(value, -1.0f, 1.0f);
-            memcpy(p, &v, sizeof(v));
-            return;
-        }
-        switch (bytes) {
-            case 2: {
-                int16_t v = (int16_t) std::clamp(
-                    (int) std::lround(value * 32768.0f), -32768, 32767);
-                memcpy(p, &v, sizeof(v));
-                break;
-            }
-            case 3: {
-                int32_t v = (int32_t) std::clamp<int64_t>(
-                    std::llround((double) value * 8388608.0), -8388608, 8388607);
-                p[0] = v & 0xFF;
-                p[1] = (v >> 8) & 0xFF;
-                p[2] = (v >> 16) & 0xFF;
-                break;
-            }
-            case 4: {
-                int32_t v = (int32_t) std::clamp<int64_t>(
-                    std::llround((double) value * 2147483648.0), INT32_MIN, INT32_MAX);
-                memcpy(p, &v, sizeof(v));
-                break;
-            }
-            default:
-                break;
-        }
-    }
-
     void Downmix::setup(const WAVEFORMATEX *game_format, WAVEFORMATEXTENSIBLE *stereo_out,
             DownmixAlgorithm algorithm) {
         this->enabled = true;
@@ -115,11 +54,7 @@ namespace hooks::audio {
         this->bytes_per_sample = game_format->wBitsPerSample / 8;
         this->game_frame_size = game_format->nChannels * this->bytes_per_sample;
 
-        // KSDATAFORMAT_SUBTYPE_IEEE_FLOAT has Data1 == 3 (matches apply_gain detection)
+        this->is_float = is_ieee_float(game_format);
-        this->is_float = game_format->wFormatTag == WAVE_FORMAT_IEEE_FLOAT
-            || (game_format->wFormatTag == WAVE_FORMAT_EXTENSIBLE
-                && reinterpret_cast<const WAVEFORMATEXTENSIBLE *>(game_format)
-                    ->SubFormat.Data1 == 0x00000003);
 
         // supported: 16/24/32-bit integer PCM and 32-bit float; anything else mixes to silence
         const bool supported = this->is_float
@@ -155,28 +90,10 @@ namespace hooks::audio {
             REFERENCE_TIME buffer_duration, REFERENCE_TIME periodicity,
             const WAVEFORMATEX *device_format, LPCGUID session_guid) {
 
-        HRESULT ret = real->Initialize(share_mode, stream_flags, buffer_duration, periodicity,
-                device_format, session_guid);
-
         // the smaller stereo buffer can end up unaligned for the device when the game sized the
-        // duration for its larger multi-channel format. recover by asking the device for the next
+        // duration for its larger multi-channel format; the helper recovers from that.
-        // aligned buffer size and re-initializing with a matching duration.
+        return initialize_with_alignment_retry(real, "audio::downmix", share_mode, stream_flags,
-        if (ret == AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED) {
+                buffer_duration, periodicity, device_format, session_guid);
-            UINT32 aligned_frames = 0;
-            if (SUCCEEDED(real->GetBufferSize(&aligned_frames)) && aligned_frames > 0) {
-                REFERENCE_TIME aligned_duration = (REFERENCE_TIME)
-                        (10000.0 * 1000 / device_format->nSamplesPerSec * aligned_frames + 0.5);
-
-                log_info("audio::downmix",
-                        "buffer not aligned, retrying with {} frames ({} hns)",
-                        aligned_frames, aligned_duration);
-
-                ret = real->Initialize(share_mode, stream_flags, aligned_duration,
-                        periodicity != 0 ? aligned_duration : 0, device_format, session_guid);
-            }
-        }
-
-        return ret;
     }
 
     void Downmix::add_channel(int channel, DWORD speaker, float gain) {
@@ -313,6 +230,21 @@ namespace hooks::audio {
         return S_OK;
     }
 
+    HRESULT Downmix::get_scratch(UINT32 frames, BYTE **ppData) {
+        const size_t needed = (size_t) frames * this->game_frame_size;
+        if (this->scratch.size() < needed) {
+            this->scratch.resize(needed);
+        }
+
+        *ppData = this->scratch.data();
+
+        return S_OK;
+    }
+
+    void Downmix::downmix_into(BYTE *dst, UINT32 frames) const {
+        this->process(dst, this->scratch.data(), frames);
+    }
+
     void Downmix::write_device_buffer(UINT32 frames, DWORD flags) {
         const int bps = this->bytes_per_sample;
         const int dst_stride = 2 * bps;

src/spice2x/hooks/audio/backends/wasapi/downmix.h

@@ -100,6 +100,14 @@ namespace hooks::audio {
         // grab the real stereo device buffer and hand the game the scratch buffer to write into.
         HRESULT get_buffer(IAudioRenderClient *real, UINT32 frames, BYTE **ppData);
 
+        // size the scratch and hand it to the game without acquiring a device buffer. used when a
+        // later stage (the resampler) owns the device interaction.
+        HRESULT get_scratch(UINT32 frames, BYTE **ppData);
+
+        // downmix the scratch the game wrote into the caller's stereo buffer, without touching the
+        // device. used to feed the resampler when the two stages are chained.
+        void downmix_into(BYTE *dst, UINT32 frames) const;
+
         // mix the scratch buffer into the stereo device buffer held since get_buffer. the caller
         // owns releasing the device buffer afterwards (see current_buffer / buffer_released).
         void write_device_buffer(UINT32 frames, DWORD flags);

src/spice2x/hooks/audio/backends/wasapi/dummy_audio_client.cpp

@@ -73,11 +73,7 @@ HRESULT STDMETHODCALLTYPE DummyIAudioClient::Initialize(
 
     // verbose output
     log_info("audio::wasapi", "IAudioClient::Initialize hook hit");
-    log_info("audio::wasapi", "... ShareMode         : {}", share_mode_str(ShareMode));
+    print_format(ShareMode, StreamFlags, hnsBufferDuration, hnsPeriodicity, pFormat);
-    log_info("audio::wasapi", "... StreamFlags       : {}", stream_flags_str(StreamFlags));
-    log_info("audio::wasapi", "... hnsBufferDuration : {}", hnsBufferDuration);
-    log_info("audio::wasapi", "... hnsPeriodicity    : {}", hnsPeriodicity);
-    print_format(pFormat);
 
     CHECK_RESULT(this->backend->on_initialize(
         &ShareMode,

src/spice2x/hooks/audio/backends/wasapi/resample.cpp

@@ -0,0 +1,437 @@
+#include "resample.h"
+
+#include <algorithm>
+#include <cmath>
+#include <cstdint>
+#include <cstring>
+#include <mutex>
+
+#include <audioclient.h>
+
+#include "util/logging.h"
+
+#include "util.h"
+
+namespace hooks::audio {
+
+    namespace {
+
+        constexpr double PI = 3.14159265358979323846;
+
+        // normalized sinc: sin(pi*x) / (pi*x), with the removable singularity at 0 filled in
+        inline double sinc(double x) {
+            if (x == 0.0) {
+                return 1.0;
+            }
+            const double px = PI * x;
+            return std::sin(px) / px;
+        }
+
+        // Blackman window across the kernel radius; zero at +/- radius
+        inline double blackman(double x, double radius) {
+            const double n = (x + radius) / (2.0 * radius);
+            if (n <= 0.0 || n >= 1.0) {
+                return 0.0;
+            }
+            return 0.42 - 0.5 * std::cos(2.0 * PI * n) + 0.08 * std::cos(4.0 * PI * n);
+        }
+    }
+
+    std::optional<uint32_t> Resampler::resolve(const WAVEFORMATEX *game_format) {
+        if (game_format == nullptr || !RESAMPLE_RATE.has_value()) {
+            return std::nullopt;
+        }
+        if (game_format->nSamplesPerSec == 0
+                || game_format->nSamplesPerSec == RESAMPLE_RATE.value()) {
+            return std::nullopt;
+        }
+        return RESAMPLE_RATE;
+    }
+
+    void Resampler::setup(const WAVEFORMATEX *game_format, WAVEFORMATEXTENSIBLE *device_out,
+            uint32_t target_rate) {
+        this->enabled = true;
+        this->channels = game_format->nChannels;
+        this->bytes_per_sample = game_format->wBitsPerSample / 8;
+        this->game_frame_size = this->channels * this->bytes_per_sample;
+
+        this->is_float = is_ieee_float(game_format);
+
+        const bool supported = this->is_float
+            ? this->bytes_per_sample == 4
+            : (this->bytes_per_sample >= 2 && this->bytes_per_sample <= 4);
+        if (!supported) {
+            log_fatal(
+                "audio::resample",
+                "unsupported sample format ({}-bit {}) for -resample",
+                game_format->wBitsPerSample, this->is_float ? "float" : "int");
+        }
+
+        this->src_rate = game_format->nSamplesPerSec;
+        this->dst_rate = target_rate;
+
+        // anti-alias cutoff: full bandwidth when upsampling, scaled down when decimating
+        this->cutoff = std::min(1.0, (double) this->dst_rate / (double) this->src_rate);
+        this->half_taps = 16;
+
+        // precompute the windowed-sinc kernel now that cutoff is known
+        this->build_kernel();
+
+        // prime the queue with half a window of silence so the first outputs have left history
+        this->in_queue.assign((size_t) this->half_taps * this->channels, 0.0f);
+        this->in_pos = this->half_taps;
+
+        this->make_device_format(game_format, device_out, target_rate);
+    }
+
+    void Resampler::make_device_format(const WAVEFORMATEX *game_format,
+            WAVEFORMATEXTENSIBLE *device_out, uint32_t target_rate) {
+        const size_t src_size = sizeof(WAVEFORMATEX) + game_format->cbSize;
+
+        memset(device_out, 0, sizeof(WAVEFORMATEXTENSIBLE));
+        memcpy(device_out, game_format, std::min(src_size, sizeof(WAVEFORMATEXTENSIBLE)));
+
+        device_out->Format.nSamplesPerSec = target_rate;
+        device_out->Format.nAvgBytesPerSec = target_rate * device_out->Format.nBlockAlign;
+    }
+
+    HRESULT Resampler::initialize(IAudioClient *real, AUDCLNT_SHAREMODE share_mode,
+            DWORD stream_flags, REFERENCE_TIME buffer_duration, REFERENCE_TIME periodicity,
+            const WAVEFORMATEX *device_format, LPCGUID session_guid) {
+
+        // the resampler bypasses the OS mixer and talks to the device directly, so it only makes
+        // sense (and only works) for exclusive streams. shared streams are already resampled by
+        // the Windows audio engine, so refuse loudly rather than silently doing nothing.
+        if (share_mode != AUDCLNT_SHAREMODE_EXCLUSIVE) {
+            log_fatal("audio::resample",
+                    "-resample requires WASAPI exclusive mode, but this stream is shared "
+                    "(Windows already resamples shared streams)");
+        }
+
+        // record the pacing model. event-driven streams fill the whole device buffer each period
+        // (produce_exact); timer-driven streams poll padding and write variable partial chunks, so
+        // they drain the pending output to the device's free space each call (flush_timer).
+        this->event_driven = (stream_flags & AUDCLNT_STREAMFLAGS_EVENTCALLBACK) != 0;
+
+        return initialize_with_alignment_retry(real, "audio::resample", share_mode, stream_flags,
+                buffer_duration, periodicity, device_format, session_guid);
+    }
+
+    UINT32 Resampler::frames_device_to_game(UINT32 device_frames) const {
+        if (this->dst_rate == 0) {
+            return device_frames;
+        }
+        // round down so the game never believes it has more room than the device can hold
+        return (UINT32) (((double) device_frames * this->src_rate) / this->dst_rate);
+    }
+
+    UINT32 Resampler::padding_device_to_game(UINT32 device_padding) const {
+        if (this->dst_rate == 0) {
+            return device_padding;
+        }
+        // round up so the reported free space stays conservative
+        return (UINT32) std::ceil(((double) device_padding * this->src_rate) / this->dst_rate);
+    }
+
+    HRESULT Resampler::get_buffer(UINT32 frames, BYTE **ppData) {
+        const size_t needed = (size_t) frames * this->game_frame_size;
+        if (this->scratch.size() < needed) {
+            this->scratch.resize(needed);
+        }
+
+        *ppData = this->scratch.data();
+
+        return S_OK;
+    }
+
+    void Resampler::enqueue_input(UINT32 frames, bool silent) {
+        const int bps = this->bytes_per_sample;
+        const int ch = this->channels;
+        const size_t base = this->in_queue.size();
+
+        this->in_queue.resize(base + (size_t) frames * ch);
+
+        if (silent || bps <= 0 || ch <= 0) {
+            std::fill(this->in_queue.begin() + base, this->in_queue.end(), 0.0f);
+            return;
+        }
+
+        const BYTE *src = this->scratch.data();
+        for (UINT32 f = 0; f < frames; f++) {
+            for (int c = 0; c < ch; c++) {
+                const size_t s = (size_t) f * ch + c;
+                this->in_queue[base + s] = read_sample(src + s * bps, bps, this->is_float);
+            }
+        }
+    }
+
+    void Resampler::build_kernel() {
+        const int taps = 2 * this->half_taps;
+        const int phases = this->kernel_phases;
+        const double cut = this->cutoff;
+        const double radius = (double) this->half_taps;
+
+        // one extra row at frac == 1.0 so emit_frame can interpolate against row p + 1 safely
+        this->kernel_table.resize((size_t) (phases + 1) * taps);
+
+        for (int p = 0; p <= phases; p++) {
+            const double frac = (double) p / (double) phases;
+            for (int k = 0; k < taps; k++) {
+                // tap k maps to input offset t = k - (half_taps - 1), matching emit_frame
+                const double x = frac - (double) (k - (this->half_taps - 1));
+                this->kernel_table[(size_t) p * taps + k] =
+                        (float) (cut * sinc(cut * x) * blackman(x, radius));
+            }
+        }
+    }
+
+    void Resampler::emit_frame() {
+        const int ch = this->channels;
+        const int radius = this->half_taps;
+        const int taps = 2 * radius;
+        const long avail = (long) (this->in_queue.size() / ch);
+        const long center = (long) std::floor(this->in_pos);
+
+        // pick the two kernel rows bracketing this fractional position and the blend between them
+        const double frac = this->in_pos - (double) center;
+        const double fp = frac * (double) this->kernel_phases;
+        const int p0 = (int) fp;
+        const float blend = (float) (fp - (double) p0);
+        const float *row0 = &this->kernel_table[(size_t) p0 * taps];
+        const float *row1 = &this->kernel_table[(size_t) (p0 + 1) * taps];
+
+        // base input index for tap 0 (t = -(radius - 1))
+        const long base = center - (radius - 1);
+
+        for (int c = 0; c < ch; c++) {
+            double acc = 0.0;
+            for (int k = 0; k < taps; k++) {
+                const long idx = base + k;
+                if (idx < 0 || idx >= avail) {
+                    continue;
+                }
+                const float w = row0[k] + blend * (row1[k] - row0[k]);
+                acc += (double) this->in_queue[(size_t) idx * ch + c] * w;
+            }
+            this->out_float.push_back((float) acc);
+        }
+    }
+
+    void Resampler::drop_consumed() {
+        const int ch = this->channels;
+        const long drop = (long) std::floor(this->in_pos) - this->half_taps;
+        if (drop > 0) {
+            const size_t drop_samples = (size_t) drop * ch;
+            if (drop_samples <= this->in_queue.size()) {
+                this->in_queue.erase(this->in_queue.begin(),
+                        this->in_queue.begin() + drop_samples);
+                this->in_pos -= drop;
+            }
+        }
+    }
+
+    UINT32 Resampler::produce_exact(UINT32 out_frames) {
+        const int ch = this->channels;
+        this->out_float.clear();
+        if (ch <= 0 || out_frames == 0) {
+            return 0;
+        }
+        this->out_float.reserve((size_t) out_frames * ch);
+
+        // resample ratio. drive it from the buffer size actually advertised to the game rather
+        // than the nominal src/dst ratio: GetBufferSize reports floor(dev_buf * src/dst) game
+        // frames, so the game only ever delivers that many input frames per device period.
+        // consuming at the nominal ratio would eat slightly more input than arrives on any device
+        // where dev_buf * src/dst is non-integer (e.g. 144 -> 132.3, floored to 132), slowly
+        // draining the queue until it underruns to permanent silence. using the advertised integer
+        // ratio keeps input and output exactly balanced; the resulting pitch error is below 0.3%
+        // and inaudible, and it collapses to the exact ratio when the division is integer (160 ->
+        // 147 stays 147/160 = 44100/48000).
+        const double step = (double) this->frames_device_to_game(this->device_buffer_frames)
+                / (double) this->device_buffer_frames;
+
+        // input frames the block will touch: from in_pos through the right edge of the sinc kernel
+        // at the final output sample. if the queue is short of this, the kernel tail reads past the
+        // end and distorts every buffer, so buffer one extra block of input before the first output
+        // (emitting silence without consuming) to build a cushion the kernel can always reach into.
+        const long avail = (long) (this->in_queue.size() / ch);
+        const long need = (long) std::ceil(this->in_pos + step * (double) out_frames)
+                + this->half_taps;
+
+        if (this->priming) {
+            if (avail < need + (long) out_frames) {
+                this->out_float.assign((size_t) out_frames * ch, 0.0f);
+                return out_frames;
+            }
+            this->priming = false;
+        }
+
+        for (UINT32 o = 0; o < out_frames; o++) {
+            this->emit_frame();
+            this->in_pos += step;
+        }
+
+        this->drop_consumed();
+        return out_frames;
+    }
+
+    UINT32 Resampler::produce_variable() {
+        const int ch = this->channels;
+        if (ch <= 0) {
+            return 0;
+        }
+
+        // input frames consumed per output frame. timer-driven streams write variable partial
+        // chunks, so produce however many output frames the currently queued input can fully
+        // support and leave the rest for the next call; this keeps input and output balanced at
+        // the exact src/dst ratio over time without depending on the device buffer size.
+        const double step = (double) this->src_rate / (double) this->dst_rate;
+        const long avail = (long) (this->in_queue.size() / ch);
+
+        // emit only while the sinc kernel's right edge stays within the queued input. the kernel
+        // reaches from in_pos out to half_taps frames ahead, so stop once that would read past the
+        // end; the remaining input becomes the next block's lookahead.
+        UINT32 produced = 0;
+        while ((long) std::ceil(this->in_pos) + this->half_taps < avail) {
+            this->emit_frame();
+            this->in_pos += step;
+            produced++;
+        }
+
+        this->drop_consumed();
+        return produced;
+    }
+
+    void Resampler::write_output(BYTE *dst, UINT32 frames, float gain) const {
+        const int bps = this->bytes_per_sample;
+        const int ch = this->channels;
+        const size_t count = (size_t) frames * ch;
+
+        for (size_t i = 0; i < count; i++) {
+            write_sample(dst + i * bps, bps, this->is_float, this->out_float[i] * gain);
+        }
+    }
+
+    HRESULT Resampler::flush(IAudioRenderClient *real, IAudioClient *client, UINT32 frames,
+            DWORD flags, float boost) {
+        if (!this->enabled) {
+            return S_OK;
+        }
+
+        // cache the device buffer size once
+        if (this->device_buffer_frames == 0) {
+            client->GetBufferSize(&this->device_buffer_frames);
+        }
+        if (this->device_buffer_frames == 0) {
+            return S_OK;
+        }
+
+        const bool silent = (flags & AUDCLNT_BUFFERFLAGS_SILENT) != 0;
+        this->enqueue_input(frames, silent);
+
+        // confirm once that conversion actually started producing output
+        static std::once_flag active_printed;
+        std::call_once(active_printed, [this]() {
+            log_info("audio::resample", "resample active: {} Hz -> {} Hz ({} ch, {})",
+                    this->src_rate, this->dst_rate, this->channels,
+                    this->event_driven ? "event-driven" : "timer-driven");
+        });
+
+        // the boost is applied here (inside write_output) rather than in the standard ReleaseBuffer
+        // path, so log it once for parity with that path's "volume boost active" line.
+        if (boost != 1.0f) {
+            static std::once_flag boost_printed;
+            std::call_once(boost_printed, [boost]() {
+                log_info("audio::resample", "volume boost active (resample): gain={}", boost);
+            });
+        }
+
+        return this->event_driven
+                ? this->flush_event(real, boost)
+                : this->flush_timer(real, client, boost);
+    }
+
+    HRESULT Resampler::flush_event(IAudioRenderClient *real, float boost) {
+
+        // event-driven exclusive streams must hand the device a full buffer every period and may
+        // not push partial counts. resample the whole input block into exactly the device buffer
+        // size.
+        const UINT32 produced = this->produce_exact(this->device_buffer_frames);
+        if (produced == 0) {
+            return S_OK;
+        }
+
+        BYTE *dev = nullptr;
+        HRESULT ret = real->GetBuffer(produced, &dev);
+        if (FAILED(ret) || dev == nullptr) {
+            return ret;
+        }
+
+        // mute the first few buffers to avoid a pop on stream start
+        float gain = boost;
+        if (this->buffers_to_mute > 0) {
+            gain = 0.0f;
+            this->buffers_to_mute--;
+        }
+
+        this->write_output(dev, produced, gain);
+
+        return real->ReleaseBuffer(produced, 0);
+    }
+
+    HRESULT Resampler::flush_timer(IAudioRenderClient *real, IAudioClient *client, float boost) {
+
+        // convert everything currently queued into the pending output FIFO (out_float). timer-
+        // driven games write variable partial chunks, so produce only what the queued input can
+        // fully support and keep the remainder for the next call.
+        this->produce_variable();
+
+        const int ch = this->channels;
+        if (ch <= 0) {
+            return S_OK;
+        }
+
+        const UINT32 pending = (UINT32) (this->out_float.size() / ch);
+        if (pending == 0) {
+            return S_OK;
+        }
+
+        // push as many frames as the device currently has free, keeping the rest queued for the
+        // next call. timer-driven games poll padding and write whenever there is room, so matching
+        // the device's free space here avoids overflowing the ring while staying device-paced.
+        UINT32 padding = 0;
+        if (FAILED(client->GetCurrentPadding(&padding))) {
+            return S_OK;
+        }
+        const UINT32 device_free = this->device_buffer_frames > padding
+                ? this->device_buffer_frames - padding
+                : 0;
+        if (device_free == 0) {
+            return S_OK;
+        }
+
+        const UINT32 to_write = std::min(pending, device_free);
+
+        BYTE *dev = nullptr;
+        HRESULT ret = real->GetBuffer(to_write, &dev);
+        if (FAILED(ret) || dev == nullptr) {
+            return ret;
+        }
+
+        // mute the first few buffers to avoid a pop on stream start
+        float gain = boost;
+        if (this->buffers_to_mute > 0) {
+            gain = 0.0f;
+            this->buffers_to_mute--;
+        }
+
+        this->write_output(dev, to_write, gain);
+        ret = real->ReleaseBuffer(to_write, 0);
+
+        // drop the frames just written from the front of the pending FIFO
+        this->out_float.erase(this->out_float.begin(),
+                this->out_float.begin() + (size_t) to_write * ch);
+
+        return ret;
+    }
+}

src/spice2x/hooks/audio/backends/wasapi/resample.h

@@ -0,0 +1,149 @@
+#pragma once
+
+#include <cstdint>
+#include <optional>
+#include <vector>
+
+#include <windows.h>
+#include <mmreg.h>
+#include <audioclient.h>
+
+#include "hooks/audio/audio.h"
+
+struct IAudioClient;
+struct IAudioRenderClient;
+
+namespace hooks::audio {
+
+    // Streaming sample-rate converter for the WASAPI render path. The real device is opened at the
+    // target rate while the game keeps writing its native-rate audio into a scratch buffer; on
+    // release that buffer is converted with a windowed-sinc kernel and pushed to the device.
+    // Channel count and sample format are preserved; only the sample rate changes.
+    //
+    // Frame counts differ between the two rates, so unlike the per-frame downmix this is stateful:
+    // a fractional read position and a window of input history carry across ReleaseBuffer calls,
+    // and the device buffer is only filled up to the space the device currently has free.
+    struct Resampler {
+
+        // whether the resampler is active for the current stream
+        bool enabled = false;
+
+        // whether the stream is event-driven (AUDCLNT_STREAMFLAGS_EVENTCALLBACK). timer-driven
+        // streams instead poll padding and write variable partial chunks, so they drain the
+        // pending output to the device's free space rather than pushing a full buffer per period.
+        bool event_driven = true;
+
+        // decide whether the stream should be resampled and to which rate. returns the target rate
+        // when RESAMPLE_RATE is set and differs from the game's rate, otherwise nullopt.
+        static std::optional<uint32_t> resolve(const WAVEFORMATEX *game_format);
+
+        // enable resampling for game_format and fill device_out with the equivalent format at the
+        // target rate to open the real device with.
+        void setup(const WAVEFORMATEX *game_format, WAVEFORMATEXTENSIBLE *device_out,
+                   uint32_t target_rate);
+
+        // build the device format equivalent to game_format at target_rate (same channels/depth).
+        static void make_device_format(const WAVEFORMATEX *game_format,
+                                       WAVEFORMATEXTENSIBLE *device_out, uint32_t target_rate);
+
+        // initialize the real device at the target rate, performing the standard WASAPI buffer
+        // realignment retry on AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED.
+        HRESULT initialize(IAudioClient *real, AUDCLNT_SHAREMODE share_mode, DWORD stream_flags,
+                           REFERENCE_TIME buffer_duration, REFERENCE_TIME periodicity,
+                           const WAVEFORMATEX *device_format, LPCGUID session_guid);
+
+        // translate a device-rate frame count to the equivalent game-rate count, so the buffer-size
+        // and padding values reported to the game stay paced at the game's native rate.
+        UINT32 frames_device_to_game(UINT32 device_frames) const;
+        UINT32 padding_device_to_game(UINT32 device_padding) const;
+
+        // hand the game a scratch buffer sized for `frames` of its native format to write into.
+        HRESULT get_buffer(UINT32 frames, BYTE **ppData);
+
+        // pointer to the input scratch (sized by get_buffer). when chained after the downmix, the
+        // downmix writes its stereo output here for the resampler to consume on the next flush.
+        BYTE *input_data() { return this->scratch.data(); }
+
+        // convert the `frames` the game wrote and push output to the real render client. `boost`
+        // is applied to the converted output. event-driven streams fill exactly one device buffer
+        // per period; timer-driven streams push as many converted frames as the device has free.
+        HRESULT flush(IAudioRenderClient *real, IAudioClient *client, UINT32 frames, DWORD flags,
+                      float boost);
+
+    private:
+
+        // append `frames` of the scratch buffer (native format), or silence, to the input queue
+        void enqueue_input(UINT32 frames, bool silent);
+
+        // event-driven path: produce exactly one full device buffer and push it.
+        HRESULT flush_event(IAudioRenderClient *real, float boost);
+
+        // timer-driven path: convert all queued input into the pending output FIFO, then push as
+        // many frames as the device currently has free, keeping the remainder for the next call.
+        HRESULT flush_timer(IAudioRenderClient *real, IAudioClient *client, float boost);
+
+        // produce exactly out_frames output frames using the fixed src/dst ratio. event-driven
+        // exclusive streams must fill the whole device buffer every period; a small input cushion
+        // is buffered first (see priming) so the sinc kernel always has lookahead.
+        UINT32 produce_exact(UINT32 out_frames);
+
+        // convert all input the kernel can fully support into the pending output FIFO (out_float),
+        // appending without clearing. returns the number of frames produced. used by the
+        // timer-driven path where output is drained to the device in device-paced chunks.
+        UINT32 produce_variable();
+
+        // convolve the windowed-sinc kernel at the current in_pos and append the resulting frame
+        // (one sample per channel) to out_float
+        void emit_frame();
+
+        // precompute the windowed-sinc kernel sampled at kernel_phases sub-sample positions, so
+        // emit_frame is a table lookup instead of recomputing sin/cos per tap (which is far too
+        // expensive to run per sample on the audio callback thread and causes underrun crackle).
+        void build_kernel();
+
+        // drop input frames that in_pos has advanced past, keeping a window of history for the
+        // next block's left context
+        void drop_consumed();
+
+        // convert the first `frames` of out_float to the device format, scaled by `gain`
+        void write_output(BYTE *dst, UINT32 frames, float gain) const;
+
+        // sample format of the stream
+        int channels = 0;
+        int bytes_per_sample = 0;
+        bool is_float = false;
+        int game_frame_size = 0;
+
+        uint32_t src_rate = 0;
+        uint32_t dst_rate = 0;
+
+        // sinc low-pass cutoff (1.0 when upsampling, dst/src when downsampling) and window radius
+        double cutoff = 1.0;
+        int half_taps = 16;
+
+        // precomputed kernel: (kernel_phases + 1) rows of 2*half_taps weights, indexed by the
+        // fractional sample position (linearly interpolated between adjacent rows in emit_frame)
+        std::vector<float> kernel_table;
+        int kernel_phases = 1024;
+
+        // interleaved float input queue and the fractional read position within it (in frames)
+        std::vector<float> in_queue;
+        double in_pos = 0.0;
+
+        // emit silence until a full block of input lookahead has accumulated, so the sinc kernel
+        // never reads past the end of the queue (which would distort the tail of every buffer)
+        bool priming = true;
+
+        // interleaved float scratch for produced output
+        std::vector<float> out_float;
+
+        // buffer the game writes its native-rate audio into between get_buffer / flush
+        std::vector<BYTE> scratch;
+
+        // cached device buffer size (frames); a full buffer is produced every period
+        UINT32 device_buffer_frames = 0;
+
+        // leading buffers to silence to avoid a pop on stream start
+        int buffers_to_mute = 16;
+    };
+}

src/spice2x/hooks/audio/backends/wasapi/util.cpp

@@ -2,7 +2,9 @@
 
 #include <audioclient.h>
 
+#include "hooks/audio/util.h"
 #include "util/flags_helper.h"
+#include "util/logging.h"
 
 #include "defs.h"
 
@@ -18,3 +20,46 @@ std::string stream_flags_str(DWORD flags) {
     FLAG(flags, AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY);
     FLAGS_END(flags);
 }
+
+void print_format(AUDCLNT_SHAREMODE share_mode, DWORD stream_flags, REFERENCE_TIME buffer_duration,
+        REFERENCE_TIME periodicity, const WAVEFORMATEX *device_format) {
+    log_info("audio::wasapi", "... ShareMode         : {}", share_mode_str(share_mode));
+    log_info("audio::wasapi", "... StreamFlags       : {}", stream_flags_str(stream_flags));
+    log_info("audio::wasapi", "... hnsBufferDuration : {} ({:.3f} ms)",
+            buffer_duration, buffer_duration / 10000.0);
+    log_info("audio::wasapi", "... hnsPeriodicity    : {} ({:.3f} ms)",
+            periodicity, periodicity / 10000.0);
+    print_format(device_format);
+}
+
+void print_format(AUDCLNT_SHAREMODE share_mode, const WAVEFORMATEX *device_format) {
+    log_info("audio::wasapi", "... ShareMode         : {}", share_mode_str(share_mode));
+    print_format(device_format);
+}
+
+HRESULT initialize_with_alignment_retry(IAudioClient *client, const char *log_group,
+        AUDCLNT_SHAREMODE share_mode, DWORD stream_flags, REFERENCE_TIME buffer_duration,
+        REFERENCE_TIME periodicity, const WAVEFORMATEX *device_format, LPCGUID session_guid) {
+
+    HRESULT ret = client->Initialize(share_mode, stream_flags, buffer_duration, periodicity,
+            device_format, session_guid);
+
+    // the requested buffer size can end up unaligned for the device; recover by asking for the next
+    // aligned buffer size and re-initializing with a matching duration.
+    if (ret == AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED) {
+        UINT32 aligned_frames = 0;
+        if (SUCCEEDED(client->GetBufferSize(&aligned_frames)) && aligned_frames > 0) {
+            REFERENCE_TIME aligned_duration = (REFERENCE_TIME)
+                    (10000.0 * 1000 / device_format->nSamplesPerSec * aligned_frames + 0.5);
+
+            log_info(log_group, "buffer not aligned, retrying with {} frames ({} hns)",
+                    aligned_frames, aligned_duration);
+
+            ret = client->Initialize(share_mode, stream_flags, aligned_duration,
+                    periodicity != 0 ? aligned_duration : 0, device_format, session_guid);
+        }
+    }
+
+    return ret;
+}
+

src/spice2x/hooks/audio/backends/wasapi/util.h

@@ -1,8 +1,100 @@
 #pragma once
 
+#include <algorithm>
+#include <cmath>
+#include <cstdint>
+#include <cstring>
 #include <string>
 
 #include <windows.h>
 #include <mmreg.h>
+#include <audioclient.h>
 
 std::string stream_flags_str(DWORD flags);
+
+// log the stream parameters (share mode, flags, buffer duration, periodicity) followed by the wave
+// format, matching the block printed at the top of IAudioClient::Initialize.
+void print_format(AUDCLNT_SHAREMODE share_mode, DWORD stream_flags, REFERENCE_TIME buffer_duration,
+        REFERENCE_TIME periodicity, const WAVEFORMATEX *device_format);
+
+// log the share mode followed by the wave format, for paths that only have a share mode (e.g.
+// IAudioClient::IsFormatSupported).
+void print_format(AUDCLNT_SHAREMODE share_mode, const WAVEFORMATEX *device_format);
+
+// detect IEEE float samples: WAVE_FORMAT_IEEE_FLOAT, or WAVE_FORMAT_EXTENSIBLE whose SubFormat is
+// KSDATAFORMAT_SUBTYPE_IEEE_FLOAT (Data1 == 3; _PCM has Data1 == 1)
+inline bool is_ieee_float(const WAVEFORMATEX *fmt) {
+    return fmt->wFormatTag == WAVE_FORMAT_IEEE_FLOAT
+        || (fmt->wFormatTag == WAVE_FORMAT_EXTENSIBLE
+            && reinterpret_cast<const WAVEFORMATEXTENSIBLE *>(fmt)->SubFormat.Data1 == 0x00000003);
+}
+
+// read one sample at `p` as a normalized float in [-1, 1]
+inline float read_sample(const BYTE *p, int bytes, bool is_float) {
+    if (is_float) {
+        float v;
+        memcpy(&v, p, sizeof(float));
+        return v;
+    }
+    switch (bytes) {
+        case 2: {
+            int16_t v;
+            memcpy(&v, p, sizeof(v));
+            return v * (1.0f / 32768.0f);
+        }
+        case 3: {
+            int32_t v = p[0] | (p[1] << 8) | (p[2] << 16);
+            if (v & 0x800000) {
+                v |= ~0xFFFFFF; // sign extend
+            }
+            return v * (1.0f / 8388608.0f);
+        }
+        case 4: {
+            int32_t v;
+            memcpy(&v, p, sizeof(v));
+            return (float) (v * (1.0 / 2147483648.0));
+        }
+        default:
+            return 0.0f;
+    }
+}
+
+// write the normalized float `value` to the sample at `p`, clamping to the format's range
+inline void write_sample(BYTE *p, int bytes, bool is_float, float value) {
+    if (is_float) {
+        float v = std::clamp(value, -1.0f, 1.0f);
+        memcpy(p, &v, sizeof(v));
+        return;
+    }
+    switch (bytes) {
+        case 2: {
+            int16_t v = (int16_t) std::clamp(
+                (int) std::lround(value * 32768.0f), -32768, 32767);
+            memcpy(p, &v, sizeof(v));
+            break;
+        }
+        case 3: {
+            int32_t v = (int32_t) std::clamp<int64_t>(
+                std::llround((double) value * 8388608.0), -8388608, 8388607);
+            p[0] = v & 0xFF;
+            p[1] = (v >> 8) & 0xFF;
+            p[2] = (v >> 16) & 0xFF;
+            break;
+        }
+        case 4: {
+            int32_t v = (int32_t) std::clamp<int64_t>(
+                std::llround((double) value * 2147483648.0), INT32_MIN, INT32_MAX);
+            memcpy(p, &v, sizeof(v));
+            break;
+        }
+        default:
+            break;
+    }
+}
+
+// initialize the real audio client, recovering from AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED by asking the
+// device for the next aligned buffer size and re-initializing with a matching duration. log_group
+// names the subsystem in the retry log line.
+HRESULT initialize_with_alignment_retry(IAudioClient *client, const char *log_group,
+        AUDCLNT_SHAREMODE share_mode, DWORD stream_flags, REFERENCE_TIME buffer_duration,
+        REFERENCE_TIME periodicity, const WAVEFORMATEX *device_format, LPCGUID session_guid);

src/spice2x/hooks/audio/util.cpp

@@ -48,33 +48,33 @@ void copy_wave_format(WAVEFORMATEXTENSIBLE *destination, const WAVEFORMATEX *sou
 }
 
 void print_format(const WAVEFORMATEX *pFormat) {
-    log_info("audio", "Wave Format:");
+    log_info("audio::wasapi", "Wave Format:");
 
     // format specific
     if (pFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
         auto format = reinterpret_cast<const WAVEFORMATEXTENSIBLE *>(pFormat);
-        log_info("audio", "... SubFormat           : {}", guid2s(format->SubFormat));
+        log_info("audio::wasapi", "... SubFormat           : {}", guid2s(format->SubFormat));
     } else {
-        log_info("audio", "... wFormatTag          : {}", pFormat->wFormatTag);
+        log_info("audio::wasapi", "... wFormatTag          : {}", pFormat->wFormatTag);
     }
 
     // generic
-    log_info("audio", "... nChannels           : {}", pFormat->nChannels);
+    log_info("audio::wasapi", "... nChannels           : {}", pFormat->nChannels);
-    log_info("audio", "... nSamplesPerSec      : {}", pFormat->nSamplesPerSec);
+    log_info("audio::wasapi", "... nSamplesPerSec      : {}", pFormat->nSamplesPerSec);
-    log_info("audio", "... nAvgBytesPerSec     : {}", pFormat->nAvgBytesPerSec);
+    log_info("audio::wasapi", "... nAvgBytesPerSec     : {}", pFormat->nAvgBytesPerSec);
-    log_info("audio", "... nBlockAlign         : {}", pFormat->nBlockAlign);
+    log_info("audio::wasapi", "... nBlockAlign         : {}", pFormat->nBlockAlign);
-    log_info("audio", "... wBitsPerSample      : {}", pFormat->wBitsPerSample);
+    log_info("audio::wasapi", "... wBitsPerSample      : {}", pFormat->wBitsPerSample);
 
     // format specific
     if (pFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
         auto format = reinterpret_cast<const WAVEFORMATEXTENSIBLE *>(pFormat);
 
         if (pFormat->wBitsPerSample == 0) {
-            log_info("audio", "... wSamplesPerBlock    : {}", format->Samples.wSamplesPerBlock);
+            log_info("audio::wasapi", "... wSamplesPerBlock    : {}", format->Samples.wSamplesPerBlock);
         } else {
-            log_info("audio", "... wValidBitsPerSample : {}", format->Samples.wValidBitsPerSample);
+            log_info("audio::wasapi", "... wValidBitsPerSample : {}", format->Samples.wValidBitsPerSample);
         }
 
-        log_info("audio", "... dwChannelMask       : {}", channel_mask_str(format->dwChannelMask));
+        log_info("audio::wasapi", "... dwChannelMask       : {}", channel_mask_str(format->dwChannelMask));
     }
 }

src/spice2x/launcher/launcher.cpp

@@ -1106,6 +1106,18 @@ int main_implementation(int argc, char *argv[]) {
             hooks::audio::VOLUME_BOOST = (float) std::pow(10.0, decibels / 20.0);
         }
     }
+    if (options[launcher::Options::AudioResample].is_active()) {
+        const uint32_t rate = options[launcher::Options::AudioResample].value_uint32();
+        if (rate > 0) {
+            hooks::audio::RESAMPLE_RATE = rate;
+        }
+    }
+    if (options[launcher::Options::AudioExclusiveBuffer].is_active()) {
+        const uint32_t ms = options[launcher::Options::AudioExclusiveBuffer].value_uint32();
+        if (ms > 0) {
+            hooks::audio::EXCLUSIVE_BUFFER_MS = ms;
+        }
+    }
 
     if (options[launcher::Options::AudioBackend].is_active()) {
         auto &name = options[launcher::Options::AudioBackend].value_text();

src/spice2x/launcher/options.cpp

@@ -1984,6 +1984,37 @@ static const std::vector<OptionDefinition> OPTION_DEFINITIONS = {
             {"30", "+30 dB"},
         },
     },
+    {
+        // AudioResample
+        .title = "WASAPI Exclusive Mode Resampling",
+        .name = "resample",
+        .desc = "Resamples the hooked audio output to a fixed sample rate before it reaches the "
+            "device. Useful when a game requests a sample rate the device cannot output in "
+            "exclusive mode (e.g. the game wants 44100 Hz but the device is locked to 48000 Hz).\n\n"
+            "Select the TARGET sample rate (one that your audio card supports).\n\n"
+            "Will result in couple milliseconds of latency and increased CPU usage when active.",
+        .type = OptionType::Enum,
+        .category = "Audio",
+        .elements = {
+            {"44100", "44.1 kHz"},
+            {"48000", "48 kHz"},
+            {"88200", "88.2 kHz"},
+            {"96000", "96 kHz"},
+            {"176400", "176.4 kHz"},
+            {"192000", "192 kHz"},
+        },
+    },
+    {
+        // AudioExclusiveBuffer
+        .title = "WASAPI Exclusive Buffer Size",
+        .name = "exclusivebuffer",
+        .desc = "Enlarges the WASAPI exclusive-mode device buffer to the specified duration in milliseconds.\n\n"
+            "Try setting this to 10 or even 20 if you hear crackling or glitches "
+            "(at the cost of slightly increased latency).",
+        .type = OptionType::Integer,
+        .setting_name = "16",
+        .category = "Audio",
+    },
     {
         // DelayBy5Seconds
         .title = "Delay by 5 Seconds (DEPRECATED - use -sleepduration instead)",

src/spice2x/launcher/options.h

@@ -201,6 +201,8 @@ namespace launcher {
             AudioDummy,
             DownmixAudioToStereo,
             VolumeBoost,
+            AudioResample,
+            AudioExclusiveBuffer,
             DelayBy5Seconds,
             spice2x_DelayByNSeconds,
             LoadStubs,