diff --git a/CMakeLists.txt b/CMakeLists.txt
index 1de4416ee..9e77a037a 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -293,6 +293,8 @@ set(AUDIO_LIB src/core/libraries/audio/audioin.cpp
               src/core/libraries/audio/audioout_backend.h
               src/core/libraries/audio/audioout_error.h
               src/core/libraries/audio/sdl_audio_out.cpp
+              src/core/libraries/audio/openal_audio_out.cpp
+              src/core/libraries/audio/openal_manager.h
               src/core/libraries/ngs2/ngs2.cpp
               src/core/libraries/ngs2/ngs2.h
 )
diff --git a/src/core/libraries/audio/audioout_backend.h b/src/core/libraries/audio/audioout_backend.h
index 0f36f19c8..ce6e261ab 100644
--- a/src/core/libraries/audio/audioout_backend.h
+++ b/src/core/libraries/audio/audioout_backend.h
@@ -31,4 +31,9 @@ public:
     std::unique_ptr<PortBackend> Open(PortOut& port) override;
 };
 
+class OpenALAudioOut final : public AudioOutBackend {
+public:
+    std::unique_ptr<PortBackend> Open(PortOut& port) override;
+};
+
 } // namespace Libraries::AudioOut
diff --git a/src/core/libraries/audio/openal_audio_out.cpp b/src/core/libraries/audio/openal_audio_out.cpp
new file mode 100644
index 000000000..9cf903d09
--- /dev/null
+++ b/src/core/libraries/audio/openal_audio_out.cpp
@@ -0,0 +1,662 @@
+// SPDX-FileCopyrightText: Copyright 2026 shadPS4 Emulator Project
+// SPDX-License-Identifier: GPL-2.0-or-later
+
+#include <algorithm>
+#include <atomic>
+#include <cstring>
+#include <memory>
+#include <thread>
+#include <vector>
+#include <AL/al.h>
+#include <AL/alc.h>
+#include <alext.h>
+#include <queue>
+
+#include "common/config.h"
+#include "common/logging/log.h"
+#include "core/libraries/audio/audioout.h"
+#include "core/libraries/audio/audioout_backend.h"
+#include "core/libraries/audio/openal_manager.h"
+#include "core/libraries/kernel/threads.h"
+
+// SIMD support detection
+#if defined(__x86_64__) || defined(_M_X64)
+#include <immintrin.h>
+#define HAS_SSE2
+#endif
+
+namespace Libraries::AudioOut {
+
+// Volume constants
+constexpr float VOLUME_0DB = 32768.0f; // 1 << 15
+constexpr float INV_VOLUME_0DB = 1.0f / VOLUME_0DB;
+constexpr float VOLUME_EPSILON = 0.001f;
+// Timing constants
+constexpr u64 VOLUME_CHECK_INTERVAL_US = 50000; // Check every 50ms
+constexpr u64 MIN_SLEEP_THRESHOLD_US = 10;
+constexpr u64 TIMING_RESYNC_THRESHOLD_US = 100000; // Resync if >100ms behind
+
+// OpenAL constants
+constexpr ALsizei NUM_BUFFERS = 4;            // Triple buffering
+constexpr ALsizei BUFFER_QUEUE_THRESHOLD = 2; // Queue more buffers when below this
+
+// Channel positions
+enum ChannelPos : u8 {
+    FL = 0,
+    FR = 1,
+    FC = 2,
+    LF = 3,
+    SL = 4,
+    SR = 5,
+    BL = 6,
+    BR = 7,
+    STD_SL = 6,
+    STD_SR = 7,
+    STD_BL = 4,
+    STD_BR = 5
+};
+
+class OpenALPortBackend : public PortBackend {
+public:
+    explicit OpenALPortBackend(const PortOut& port)
+        : frame_size(port.format_info.FrameSize()), guest_buffer_size(port.BufferSize()),
+          buffer_frames(port.buffer_frames), sample_rate(port.sample_rate),
+          num_channels(port.format_info.num_channels), is_float(port.format_info.is_float),
+          is_std(port.format_info.is_std), channel_layout(port.format_info.channel_layout) {
+
+        if (!Initialize(port.type)) {
+            LOG_ERROR(Lib_AudioOut, "Failed to initialize OpenAL audio backend");
+        }
+    }
+
+    ~OpenALPortBackend() override {
+        Cleanup();
+    }
+
+    void Output(void* ptr) override {
+        if (!source || buffers.empty() || !convert) [[unlikely]] {
+            return;
+        }
+
+        if (ptr == nullptr) [[unlikely]] {
+            return;
+        }
+
+        // Make context current before any OpenAL operations
+        if (!device_context->MakeCurrent()) {
+            return;
+        }
+
+        UpdateVolumeIfChanged();
+        const u64 current_time = Kernel::sceKernelGetProcessTime();
+
+        // Process audio data
+        convert(ptr, al_buffer.data(), buffer_frames, nullptr);
+        HandleTiming(current_time);
+
+        // Manage buffer queue
+        while (!available_buffers.empty() && available_buffers.size() < NUM_BUFFERS) {
+            ALint processed = 0;
+            alGetSourcei(source, AL_BUFFERS_PROCESSED, &processed);
+
+            if (processed <= 0)
+                break;
+
+            ALuint buffer_id;
+            alSourceUnqueueBuffers(source, 1, &buffer_id); // One at a time
+
+            if (alGetError() != AL_NO_ERROR) {
+                break; // Stop on error
+            }
+
+            available_buffers.push_back(buffer_id);
+        }
+
+        // Check if we need to queue more buffers
+        ALint queued = 0;
+        alGetSourcei(source, AL_BUFFERS_QUEUED, &queued);
+
+        if (queued < BUFFER_QUEUE_THRESHOLD && !available_buffers.empty()) {
+            ALuint buffer_id = available_buffers.back();
+            available_buffers.pop_back();
+
+            alBufferData(buffer_id, format, al_buffer.data(), buffer_size_bytes, sample_rate);
+
+            alSourceQueueBuffers(source, 1, &buffer_id);
+            queued++;
+        }
+        // Ensure the source is playing (OpenAL does NOT auto-start)
+        ALint state = 0;
+        alGetSourcei(source, AL_SOURCE_STATE, &state);
+
+        if (state != AL_PLAYING && queued > 0) {
+            alSourcePlay(source);
+        }
+        // Update timing
+        last_output_time.store(current_time, std::memory_order_release);
+        output_count++;
+    }
+
+    void SetVolume(const std::array<int, 8>& ch_volumes) override {
+        // Make context current before any OpenAL operations
+        if (!device_context->MakeCurrent()) {
+            return;
+        }
+
+        if (!source) [[unlikely]] {
+            return;
+        }
+
+        float max_channel_gain = 0.0f;
+        const u32 channels_to_check = std::min(num_channels, 8u);
+
+        for (u32 i = 0; i < channels_to_check; i++) {
+            const float channel_gain = static_cast<float>(ch_volumes[i]) * INV_VOLUME_0DB;
+            max_channel_gain = std::max(max_channel_gain, channel_gain);
+        }
+
+        const float slider_gain = Config::getVolumeSlider() * 0.01f;
+        const float total_gain = max_channel_gain * slider_gain;
+
+        // Only update if changed significantly
+        const float current = current_gain.load(std::memory_order_acquire);
+        if (std::abs(total_gain - current) < VOLUME_EPSILON) {
+            return;
+        }
+
+        // Apply volume change to OpenAL source
+        alSourcef(source, AL_GAIN, total_gain);
+
+        ALenum error = alGetError();
+        if (error == AL_NO_ERROR) {
+            current_gain.store(total_gain, std::memory_order_release);
+            LOG_DEBUG(Lib_AudioOut,
+                      "Set combined audio gain to {:.3f} (channel: {:.3f}, slider: {:.3f})",
+                      total_gain, max_channel_gain, slider_gain);
+        } else {
+            LOG_ERROR(Lib_AudioOut, "Failed to set OpenAL source gain: {}",
+                      GetALErrorString(error));
+        }
+    }
+
+    u64 GetLastOutputTime() const {
+        return last_output_time.load(std::memory_order_acquire);
+    }
+
+private:
+    bool Initialize(OrbisAudioOutPort type) {
+        // Get OpenAL device and context
+        if (!OpenALDevice::GetInstance().IsInitialized()) {
+            LOG_ERROR(Lib_AudioOut, "OpenAL device not initialized");
+            return false;
+        }
+
+        device_context = &OpenALDevice::GetInstance();
+
+        // Make context current for initialization
+        if (!device_context->MakeCurrent()) {
+            LOG_ERROR(Lib_AudioOut, "Failed to make OpenAL context current");
+            return false;
+        }
+
+        // Calculate timing parameters
+        period_us = (1000000ULL * buffer_frames + sample_rate / 2) / sample_rate;
+
+        // Determine OpenAL format
+        if (!DetermineOpenALFormat()) {
+            LOG_ERROR(Lib_AudioOut, "Unsupported audio format for OpenAL");
+            return false;
+        }
+
+        // Calculate buffer size
+        buffer_size_bytes = buffer_frames * frame_size;
+
+        // Allocate current buffer
+        current_buffer.resize(buffer_frames * num_channels); // float staging
+        al_buffer.resize(buffer_frames * num_channels);      // int16 upload
+        buffer_size_bytes = buffer_frames * num_channels * sizeof(s16);
+
+        // Select optimal converter function
+        if (!SelectConverter()) {
+            return false;
+        }
+
+        // Generate OpenAL source and buffers
+        if (!CreateOpenALObjects()) {
+            return false;
+        }
+
+        // Initialize current gain
+        current_gain.store(Config::getVolumeSlider() * 0.01f, std::memory_order_relaxed);
+
+        // Apply initial volume
+        alSourcef(source, AL_GAIN, current_gain.load(std::memory_order_relaxed));
+
+        LOG_INFO(Lib_AudioOut, "Initialized OpenAL backend ({} Hz, {} ch, {} format)", sample_rate,
+                 num_channels, is_float ? "float" : "int16");
+
+        return true;
+    }
+
+    void Cleanup() {
+        if (!device_context->MakeCurrent()) {
+            return;
+        }
+
+        if (source) {
+            alSourceStop(source);
+
+            ALint queued = 0;
+            alGetSourcei(source, AL_BUFFERS_QUEUED, &queued);
+            while (queued-- > 0) {
+                ALuint buf;
+                alSourceUnqueueBuffers(source, 1, &buf);
+            }
+
+            alDeleteSources(1, &source);
+            source = 0;
+        }
+
+        if (!buffers.empty()) {
+            alDeleteBuffers(static_cast<ALsizei>(buffers.size()), buffers.data());
+            buffers.clear();
+        }
+    }
+
+    void UpdateVolumeIfChanged() {
+        const u64 current_time = Kernel::sceKernelGetProcessTime();
+
+        if (current_time - last_volume_check_time < VOLUME_CHECK_INTERVAL_US) {
+            return;
+        }
+
+        last_volume_check_time = current_time;
+
+        const float config_volume = Config::getVolumeSlider() * 0.01f;
+        const float stored_gain = current_gain.load(std::memory_order_acquire);
+
+        // Only update if the difference is significant
+        if (std::abs(config_volume - stored_gain) > VOLUME_EPSILON) {
+            alSourcef(source, AL_GAIN, config_volume);
+
+            ALenum error = alGetError();
+            if (error == AL_NO_ERROR) {
+                current_gain.store(config_volume, std::memory_order_release);
+                LOG_DEBUG(Lib_AudioOut, "Updated audio gain to {:.3f}", config_volume);
+            } else {
+                LOG_ERROR(Lib_AudioOut, "Failed to set audio gain: {}", GetALErrorString(error));
+            }
+        }
+    }
+
+    void HandleTiming(u64 current_time) {
+        if (next_output_time == 0) [[unlikely]] {
+            // First output - set initial timing
+            next_output_time = current_time + period_us;
+            return;
+        }
+
+        const s64 time_diff = static_cast<s64>(current_time - next_output_time);
+
+        if (time_diff > static_cast<s64>(TIMING_RESYNC_THRESHOLD_US)) [[unlikely]] {
+            // We're far behind - resync
+            next_output_time = current_time + period_us;
+        } else if (time_diff < 0) {
+            // We're ahead of schedule - wait
+            const u64 time_to_wait = static_cast<u64>(-time_diff);
+            next_output_time += period_us;
+
+            if (time_to_wait > MIN_SLEEP_THRESHOLD_US) {
+                // Sleep for most of the wait period
+                const u64 sleep_duration = time_to_wait - MIN_SLEEP_THRESHOLD_US;
+                std::this_thread::sleep_for(std::chrono::microseconds(sleep_duration));
+            }
+        } else {
+            // Slightly behind or on time - just advance
+            next_output_time += period_us;
+        }
+    }
+
+    bool DetermineOpenALFormat() {
+        if (is_float) {
+            // OpenAL doesn't natively support float formats, we need to use AL_EXT_FLOAT32
+            // extension For simplicity, we'll convert to int16 in our converter functions
+            format = AL_FORMAT_MONO16; // Default, will be overridden
+
+            switch (num_channels) {
+            case 1:
+                format = AL_FORMAT_MONO16;
+                break;
+            case 2:
+                format = AL_FORMAT_STEREO16;
+                break;
+            case 6:
+                format = alGetEnumValue("AL_FORMAT_51CHN16");
+                if (format == 0 || alGetError() != AL_NO_ERROR) {
+                    LOG_WARNING(Lib_AudioOut, "5.1 format not supported, falling back to stereo");
+                    format = AL_FORMAT_STEREO16;
+                }
+                break;
+            case 8:
+                format = alGetEnumValue("AL_FORMAT_71CHN16");
+                if (format == 0 || alGetError() != AL_NO_ERROR) {
+                    LOG_WARNING(Lib_AudioOut, "7.1 format not supported, falling back to stereo");
+                    format = AL_FORMAT_STEREO16;
+                }
+                break;
+            default:
+                LOG_ERROR(Lib_AudioOut, "Unsupported float channel count: {}", num_channels);
+                return false;
+            }
+        } else {
+            // 16-bit integer formats
+            switch (num_channels) {
+            case 1:
+                format = AL_FORMAT_MONO16;
+                break;
+            case 2:
+                format = AL_FORMAT_STEREO16;
+                break;
+            case 6:
+                format = alGetEnumValue("AL_FORMAT_51CHN16");
+                if (format == 0 || alGetError() != AL_NO_ERROR) {
+                    LOG_WARNING(Lib_AudioOut, "5.1 format not supported, falling back to stereo");
+                    format = AL_FORMAT_STEREO16;
+                }
+                break;
+            case 8:
+                format = alGetEnumValue("AL_FORMAT_71CHN16");
+                if (format == 0 || alGetError() != AL_NO_ERROR) {
+                    LOG_WARNING(Lib_AudioOut, "7.1 format not supported, falling back to stereo");
+                    format = AL_FORMAT_STEREO16;
+                }
+                break;
+            default:
+                LOG_ERROR(Lib_AudioOut, "Unsupported S16 channel count: {}", num_channels);
+                return false;
+            }
+        }
+
+        return true;
+    }
+
+    bool CreateOpenALObjects() {
+        // Generate source
+        alGenSources(1, &source);
+        if (alGetError() != AL_NO_ERROR) {
+            LOG_ERROR(Lib_AudioOut, "Failed to generate OpenAL source");
+            return false;
+        }
+
+        // Generate buffers
+        buffers.resize(NUM_BUFFERS);
+        alGenBuffers(static_cast<ALsizei>(buffers.size()), buffers.data());
+        if (alGetError() != AL_NO_ERROR) {
+            LOG_ERROR(Lib_AudioOut, "Failed to generate OpenAL buffers");
+            alDeleteSources(1, &source);
+            source = 0;
+            return false;
+        }
+
+        // All buffers are initially available
+        available_buffers = buffers;
+
+        // Configure source properties
+        alSourcef(source, AL_PITCH, 1.0f);
+        alSourcef(source, AL_GAIN, 1.0f);
+        alSource3f(source, AL_POSITION, 0.0f, 0.0f, 0.0f);
+        alSource3f(source, AL_VELOCITY, 0.0f, 0.0f, 0.0f);
+        alSourcei(source, AL_LOOPING, AL_FALSE);
+        alSourcei(source, AL_SOURCE_RELATIVE, AL_TRUE);
+
+        LOG_DEBUG(Lib_AudioOut, "Created OpenAL source {} with {} buffers", source, buffers.size());
+
+        return true;
+    }
+
+    bool SelectConverter() {
+        if (is_float) {
+            // For OpenAL, we need to convert float to int16
+            switch (num_channels) {
+            case 1:
+                convert = &ConvertF32ToS16Mono;
+                break;
+            case 2:
+                convert = &ConvertF32ToS16Stereo;
+                break;
+            case 8:
+                convert = is_std ? &ConvertF32ToS16Std8CH : &ConvertF32ToS16_8CH;
+                break;
+            default:
+                LOG_ERROR(Lib_AudioOut, "Unsupported float channel count: {}", num_channels);
+                return false;
+            }
+        } else {
+            switch (num_channels) {
+            case 1:
+                convert = &ConvertS16Mono;
+                break;
+            case 2:
+#if defined(HAS_SSE2)
+                convert = &ConvertS16StereoSIMD;
+#else
+                convert = &ConvertS16Stereo;
+#endif
+                break;
+            case 8:
+#if defined(HAS_SSE2)
+                convert = &ConvertS16_8CH_SIMD;
+#else
+                convert = &ConvertS16_8CH;
+#endif
+                break;
+            default:
+                LOG_ERROR(Lib_AudioOut, "Unsupported S16 channel count: {}", num_channels);
+                return false;
+            }
+        }
+
+        return true;
+    }
+
+    const char* GetALErrorString(ALenum error) {
+        switch (error) {
+        case AL_NO_ERROR:
+            return "AL_NO_ERROR";
+        case AL_INVALID_NAME:
+            return "AL_INVALID_NAME";
+        case AL_INVALID_ENUM:
+            return "AL_INVALID_ENUM";
+        case AL_INVALID_VALUE:
+            return "AL_INVALID_VALUE";
+        case AL_INVALID_OPERATION:
+            return "AL_INVALID_OPERATION";
+        case AL_OUT_OF_MEMORY:
+            return "AL_OUT_OF_MEMORY";
+        default:
+            return "Unknown AL error";
+        }
+    }
+
+    // Converter function type
+    using ConverterFunc = void (*)(const void* src, void* dst, u32 frames, const float* volumes);
+
+    // S16 converters (same as SDL backend)
+    static void ConvertS16Mono(const void* src, void* dst, u32 frames, const float*) {
+        const s16* s = static_cast<const s16*>(src);
+        float* d = static_cast<float*>(dst);
+
+        for (u32 i = 0; i < frames; i++) {
+            d[i] = s[i] * INV_VOLUME_0DB;
+        }
+    }
+
+    static void ConvertS16Stereo(const void* src, void* dst, u32 frames, const float*) {
+        const s16* s = static_cast<const s16*>(src);
+        float* d = static_cast<float*>(dst);
+
+        const u32 num_samples = frames << 1;
+        for (u32 i = 0; i < num_samples; i++) {
+            d[i] = s[i] * INV_VOLUME_0DB;
+        }
+    }
+
+#ifdef HAS_SSE2
+    static void ConvertS16StereoSIMD(const void* src, void* dst, u32 frames, const float*) {
+        const s16* s = static_cast<const s16*>(src);
+        float* d = static_cast<float*>(dst);
+
+        const __m128 scale = _mm_set1_ps(INV_VOLUME_0DB);
+        const u32 num_samples = frames << 1;
+        u32 i = 0;
+
+        for (; i + 8 <= num_samples; i += 8) {
+            __m128i s16_vals = _mm_loadu_si128(reinterpret_cast<const __m128i*>(&s[i]));
+            __m128i s32_lo = _mm_cvtepi16_epi32(s16_vals);
+            __m128i s32_hi = _mm_cvtepi16_epi32(_mm_srli_si128(s16_vals, 8));
+            __m128 f_lo = _mm_mul_ps(_mm_cvtepi32_ps(s32_lo), scale);
+            __m128 f_hi = _mm_mul_ps(_mm_cvtepi32_ps(s32_hi), scale);
+            _mm_storeu_ps(&d[i], f_lo);
+            _mm_storeu_ps(&d[i + 4], f_hi);
+        }
+
+        for (; i < num_samples; i++) {
+            d[i] = s[i] * INV_VOLUME_0DB;
+        }
+    }
+#endif
+
+    static void ConvertS16_8CH(const void* src, void* dst, u32 frames, const float*) {
+        const s16* s = static_cast<const s16*>(src);
+        float* d = static_cast<float*>(dst);
+
+        const u32 num_samples = frames << 3;
+        for (u32 i = 0; i < num_samples; i++) {
+            d[i] = s[i] * INV_VOLUME_0DB;
+        }
+    }
+
+#ifdef HAS_SSE2
+    static void ConvertS16_8CH_SIMD(const void* src, void* dst, u32 frames, const float*) {
+        const s16* s = static_cast<const s16*>(src);
+        float* d = static_cast<float*>(dst);
+
+        const __m128 scale = _mm_set1_ps(INV_VOLUME_0DB);
+        const u32 num_samples = frames << 3;
+        u32 i = 0;
+
+        for (; i + 8 <= num_samples; i += 8) {
+            __m128i s16_vals = _mm_loadu_si128(reinterpret_cast<const __m128i*>(&s[i]));
+            __m128i s32_lo = _mm_cvtepi16_epi32(s16_vals);
+            __m128i s32_hi = _mm_cvtepi16_epi32(_mm_srli_si128(s16_vals, 8));
+            __m128 f_lo = _mm_mul_ps(_mm_cvtepi32_ps(s32_lo), scale);
+            __m128 f_hi = _mm_mul_ps(_mm_cvtepi32_ps(s32_hi), scale);
+            _mm_storeu_ps(&d[i], f_lo);
+            _mm_storeu_ps(&d[i + 4], f_hi);
+        }
+
+        for (; i < num_samples; i++) {
+            d[i] = s[i] * INV_VOLUME_0DB;
+        }
+    }
+#endif
+
+    // Float to S16 converters for OpenAL
+    static void ConvertF32ToS16Mono(const void* src, void* dst, u32 frames, const float*) {
+        const float* s = static_cast<const float*>(src);
+        s16* d = static_cast<s16*>(dst);
+
+        for (u32 i = 0; i < frames; i++) {
+            const float sample = s[i] * VOLUME_0DB;
+            d[i] = static_cast<s16>(std::clamp(sample, -32768.0f, 32767.0f));
+        }
+    }
+
+    static void ConvertF32ToS16Stereo(const void* src, void* dst, u32 frames, const float*) {
+        const float* s = static_cast<const float*>(src);
+        s16* d = static_cast<s16*>(dst);
+
+        const u32 num_samples = frames << 1;
+        for (u32 i = 0; i < num_samples; i++) {
+            const float sample = s[i] * VOLUME_0DB;
+            d[i] = static_cast<s16>(std::clamp(sample, -32768.0f, 32767.0f));
+        }
+    }
+
+    static void ConvertF32ToS16_8CH(const void* src, void* dst, u32 frames, const float*) {
+        const float* s = static_cast<const float*>(src);
+        s16* d = static_cast<s16*>(dst);
+
+        const u32 num_samples = frames << 3;
+        for (u32 i = 0; i < num_samples; i++) {
+            const float sample = s[i] * VOLUME_0DB;
+            d[i] = static_cast<s16>(std::clamp(sample, -32768.0f, 32767.0f));
+        }
+    }
+
+    static void ConvertF32ToS16Std8CH(const void* src, void* dst, u32 frames, const float*) {
+        const float* s = static_cast<const float*>(src);
+        s16* d = static_cast<s16*>(dst);
+
+        for (u32 i = 0; i < frames; i++) {
+            const u32 offset = i << 3;
+
+            d[offset + FL] =
+                static_cast<s16>(std::clamp(s[offset + FL] * VOLUME_0DB, -32768.0f, 32767.0f));
+            d[offset + FR] =
+                static_cast<s16>(std::clamp(s[offset + FR] * VOLUME_0DB, -32768.0f, 32767.0f));
+            d[offset + FC] =
+                static_cast<s16>(std::clamp(s[offset + FC] * VOLUME_0DB, -32768.0f, 32767.0f));
+            d[offset + LF] =
+                static_cast<s16>(std::clamp(s[offset + LF] * VOLUME_0DB, -32768.0f, 32767.0f));
+            d[offset + SL] =
+                static_cast<s16>(std::clamp(s[offset + STD_SL] * VOLUME_0DB, -32768.0f, 32767.0f));
+            d[offset + SR] =
+                static_cast<s16>(std::clamp(s[offset + STD_SR] * VOLUME_0DB, -32768.0f, 32767.0f));
+            d[offset + BL] =
+                static_cast<s16>(std::clamp(s[offset + STD_BL] * VOLUME_0DB, -32768.0f, 32767.0f));
+            d[offset + BR] =
+                static_cast<s16>(std::clamp(s[offset + STD_BR] * VOLUME_0DB, -32768.0f, 32767.0f));
+        }
+    }
+
+    // Audio format parameters
+    const u32 frame_size;
+    const u32 guest_buffer_size;
+    const u32 buffer_frames;
+    const u32 sample_rate;
+    const u32 num_channels;
+    const bool is_float;
+    const bool is_std;
+    const std::array<int, 8> channel_layout;
+
+    alignas(64) u64 period_us{0};
+    alignas(64) std::atomic<u64> last_output_time{0};
+    u64 next_output_time{0};
+    u64 last_volume_check_time{0};
+    u32 output_count{0};
+
+    // OpenAL objects
+    OpenALDevice* device_context{nullptr};
+    ALuint source{0};
+    std::vector<ALuint> buffers;
+    std::vector<ALuint> available_buffers;
+    ALenum format{AL_FORMAT_STEREO16};
+
+    // Buffer management
+    u32 buffer_size_bytes{0};
+    std::vector<float> current_buffer;
+    std::vector<s16> al_buffer;
+
+    // Converter function pointer
+    ConverterFunc convert{nullptr};
+
+    // Volume management
+    alignas(64) std::atomic<float> current_gain{1.0f};
+};
+
+std::unique_ptr<PortBackend> OpenALAudioOut::Open(PortOut& port) {
+    return std::make_unique<OpenALPortBackend>(port);
+}
+
+} // namespace Libraries::AudioOut
\ No newline at end of file
diff --git a/src/core/libraries/audio/openal_manager.h b/src/core/libraries/audio/openal_manager.h
new file mode 100644
index 000000000..4f58fca18
--- /dev/null
+++ b/src/core/libraries/audio/openal_manager.h
@@ -0,0 +1,110 @@
+// SPDX-FileCopyrightText: Copyright 2026 shadPS4 Emulator Project
+// SPDX-License-Identifier: GPL-2.0-or-later
+
+#pragma once
+
+#include <mutex>
+#include <AL/al.h>
+#include <AL/alc.h>
+
+namespace Libraries::AudioOut {
+
+class OpenALDevice {
+public:
+    static OpenALDevice& GetInstance() {
+        static OpenALDevice instance;
+        return instance;
+    }
+
+    bool IsInitialized() const {
+        return initialized;
+    }
+
+    ALCdevice* GetDevice() const {
+        return device;
+    }
+
+    ALCcontext* GetContext() const {
+        return context;
+    }
+
+    bool MakeCurrent() {
+        std::lock_guard<std::mutex> lock(mutex);
+
+        if (!initialized) {
+            return false;
+        }
+
+        return alcMakeContextCurrent(context);
+    }
+
+    void ReleaseContext() {
+        std::lock_guard<std::mutex> lock(mutex);
+        alcMakeContextCurrent(nullptr);
+    }
+
+private:
+    OpenALDevice() {
+        Initialize();
+    }
+
+    ~OpenALDevice() {
+        Cleanup();
+    }
+
+    OpenALDevice(const OpenALDevice&) = delete;
+    OpenALDevice& operator=(const OpenALDevice&) = delete;
+
+    void Initialize() {
+        std::lock_guard<std::mutex> lock(mutex);
+
+        if (initialized) {
+            return;
+        }
+
+        // Open default device
+        device = alcOpenDevice(nullptr);
+        if (!device) {
+            return;
+        }
+
+        // Create context
+        context = alcCreateContext(device, nullptr);
+        if (!context) {
+            alcCloseDevice(device);
+            device = nullptr;
+            return;
+        }
+
+        initialized = true;
+    }
+
+    void Cleanup() {
+        std::lock_guard<std::mutex> lock(mutex);
+
+        if (!initialized) {
+            return;
+        }
+
+        ReleaseContext();
+
+        if (context) {
+            alcDestroyContext(context);
+            context = nullptr;
+        }
+
+        if (device) {
+            alcCloseDevice(device);
+            device = nullptr;
+        }
+
+        initialized = false;
+    }
+
+    ALCdevice* device{nullptr};
+    ALCcontext* context{nullptr};
+    bool initialized{false};
+    mutable std::mutex mutex;
+};
+
+} // namespace Libraries::AudioOut
\ No newline at end of file