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https://github.com/shadps4-emu/shadPS4.git
synced 2026-04-09 11:11:29 -06:00
more safe format conversions
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georgemoralis
parent
d06491353d
commit
f6e46f18df
@ -1,4 +1,4 @@
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// SPDX-FileCopyrightText: Copyright 2025 shadPS4 Emulator Project
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// SPDX-FileCopyrightText: Copyright 2025 shadPS4 Emulator Project
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// SPDX-License-Identifier: GPL-2.0-or-later
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#include <algorithm>
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@ -80,104 +80,165 @@ s32 PS4_SYSV_ABI sceAudio3dAudioOutOutputs(AudioOut::OrbisAudioOutOutputParam* p
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return AudioOut::sceAudioOutOutputs(param, num);
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}
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static inline s16 OrbisFloatToS16(float v) {
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if (std::abs(v) < 1.0e-20f)
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v = 0.0f;
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// Sony behavior: +1.0f -> 32767, -1.0f -> -32768
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const float scaled = v * 32768.0f;
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if (scaled >= 32767.0f)
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return 32767;
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if (scaled <= -32768.0f)
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return -32768;
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return static_cast<s16>(scaled + (scaled >= 0 ? 0.5f : -0.5f));
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}
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static void ConvertAudioToStereoS16(const OrbisAudio3dPcm& pcm, const u32 num_channels,
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u8** out_data, u32* out_size) {
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const u32 num_samples = pcm.num_samples;
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const u32 output_samples = num_samples * 2; // Stereo output
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*out_size = output_samples * sizeof(s16);
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*out_data = static_cast<u8*>(malloc(*out_size));
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*out_data = nullptr;
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*out_size = 0;
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if (!*out_data) {
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*out_size = 0;
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// Validate input parameters
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if (!pcm.sample_buffer || pcm.num_samples == 0 || num_channels == 0) {
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return;
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}
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s16* dst = reinterpret_cast<s16*>(*out_data);
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// Check for overflow in size calculation
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const u32 num_frames = pcm.num_samples;
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if (pcm.format == OrbisAudio3dFormat::ORBIS_AUDIO3D_FORMAT_S16) {
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const s16* src = static_cast<const s16*>(pcm.sample_buffer);
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// Validate frame count (prevent excessive allocation)
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constexpr u32 MAX_FRAMES = 48000 * 10; // 10 seconds at 48kHz
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if (num_frames > MAX_FRAMES || num_frames == 0) {
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return;
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}
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if (num_channels == 2) {
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// Already stereo S16, just copy
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std::memcpy(dst, src, *out_size);
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} else if (num_channels == 8) {
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// For 3D audio, just take L and R channels (0 and 1)
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// The game should have mixed all 3D positioning into these
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for (u32 i = 0; i < num_samples; i++) {
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const s16* frame = &src[i * 8];
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dst[i * 2 + 0] = frame[0]; // L
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dst[i * 2 + 1] = frame[1]; // R
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// Calculate output size with overflow checking
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if (num_frames > (std::numeric_limits<u32>::max() / 2)) {
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return; // Overflow in out_samples calculation
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}
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const u32 out_samples = num_frames * 2; // stereo
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if (out_samples > (std::numeric_limits<u32>::max() / sizeof(s16))) {
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return; // Overflow in out_bytes calculation
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}
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const u32 out_bytes = out_samples * sizeof(s16);
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// Allocate output buffer
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s16* dst = static_cast<s16*>(std::malloc(out_bytes));
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if (!dst) {
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return;
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}
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// Initialize to silence
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std::memset(dst, 0, out_bytes);
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*out_data = reinterpret_cast<u8*>(dst);
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*out_size = out_bytes;
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// Calculate expected input size for validation
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const size_t sample_size =
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(pcm.format == OrbisAudio3dFormat::ORBIS_AUDIO3D_FORMAT_S16) ? sizeof(s16) : sizeof(float);
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const size_t expected_input_bytes = num_frames * num_channels * sample_size;
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try {
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switch (pcm.format) {
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case OrbisAudio3dFormat::ORBIS_AUDIO3D_FORMAT_S16: {
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const s16* src = static_cast<const s16*>(pcm.sample_buffer);
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switch (num_channels) {
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case 2:
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std::memcpy(dst, src, out_bytes);
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break;
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case 1:
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// Mono to stereo
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for (u32 i = 0; i < num_frames; ++i) {
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if (i >= num_frames)
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break;
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s16 s = src[i];
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dst[i * 2 + 0] = s;
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dst[i * 2 + 1] = s;
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}
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break;
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case 8:
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for (u32 i = 0; i < num_frames; ++i) {
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if (i * 8 >= num_frames * num_channels)
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break; // Prevent overflow
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const s16* frame = &src[i * 8];
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dst[i * 2 + 0] = frame[0];
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dst[i * 2 + 1] = frame[1];
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}
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break;
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default:
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// Unsupported channel layout - already memset to 0
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LOG_ERROR(Lib_Audio3d, "Unsupported S16 channel count: {}", num_channels);
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break;
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}
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} else if (num_channels == 1) {
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// Mono to stereo
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for (u32 i = 0; i < num_samples; i++) {
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s16 sample = src[i];
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dst[i * 2 + 0] = sample;
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dst[i * 2 + 1] = sample;
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}
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} else {
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std::memset(dst, 0, *out_size);
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break;
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}
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} else if (pcm.format == OrbisAudio3dFormat::ORBIS_AUDIO3D_FORMAT_FLOAT) {
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const float* src = static_cast<const float*>(pcm.sample_buffer);
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constexpr float SCALE = 32767.0f;
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if (num_channels == 2) {
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// Stereo float to stereo S16
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for (u32 i = 0; i < num_samples; i++) {
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float left = src[i * 2 + 0];
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float right = src[i * 2 + 1];
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case OrbisAudio3dFormat::ORBIS_AUDIO3D_FORMAT_FLOAT: {
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const float* src = static_cast<const float*>(pcm.sample_buffer);
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// Soft clipping to prevent harsh distortion
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if (std::abs(left) > 1.0f) {
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left = std::copysign(1.0f - 0.5f / std::abs(left), left);
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}
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if (std::abs(right) > 1.0f) {
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right = std::copysign(1.0f - 0.5f / std::abs(right), right);
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switch (num_channels) {
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case 2:
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// Stereo float to stereo S16
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for (u32 i = 0; i < num_frames; ++i) {
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if (i >= num_frames)
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break;
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dst[i * 2 + 0] = OrbisFloatToS16(src[i * 2 + 0]);
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dst[i * 2 + 1] = OrbisFloatToS16(src[i * 2 + 1]);
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}
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break;
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dst[i * 2 + 0] = static_cast<s16>(left * SCALE);
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dst[i * 2 + 1] = static_cast<s16>(right * SCALE);
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case 1:
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// Mono float to stereo
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for (u32 i = 0; i < num_frames; ++i) {
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if (i >= num_frames)
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break;
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s16 s = OrbisFloatToS16(src[i]);
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dst[i * 2 + 0] = s;
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dst[i * 2 + 1] = s;
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}
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break;
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case 8:
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// Audio3D float to stereo (L/R only)
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for (u32 i = 0; i < num_frames; ++i) {
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if (i * 8 >= num_frames * num_channels)
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break;
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const float* frame = &src[i * 8];
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dst[i * 2 + 0] = OrbisFloatToS16(frame[0]);
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dst[i * 2 + 1] = OrbisFloatToS16(frame[1]);
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}
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break;
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default:
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// Already memset to 0
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LOG_ERROR(Lib_Audio3d, "Unsupported float channel count: {}", num_channels);
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break;
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}
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} else if (num_channels == 8) {
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// For 3D audio, the game mixes everything into L and R
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// Just take channels 0 and 1
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for (u32 i = 0; i < num_samples; i++) {
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const float* frame = &src[i * 8];
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float left = frame[0];
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float right = frame[1];
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// Soft clipping
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if (std::abs(left) > 1.0f) {
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left = std::copysign(1.0f - 0.5f / std::abs(left), left);
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}
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if (std::abs(right) > 1.0f) {
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right = std::copysign(1.0f - 0.5f / std::abs(right), right);
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}
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dst[i * 2 + 0] = static_cast<s16>(left * SCALE);
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dst[i * 2 + 1] = static_cast<s16>(right * SCALE);
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}
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} else if (num_channels == 1) {
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// Mono float to stereo
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for (u32 i = 0; i < num_samples; i++) {
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float sample = src[i];
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if (std::abs(sample) > 1.0f) {
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sample = std::copysign(1.0f - 0.5f / std::abs(sample), sample);
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}
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s16 converted = static_cast<s16>(sample * SCALE);
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dst[i * 2 + 0] = converted;
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dst[i * 2 + 1] = converted;
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}
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} else {
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std::memset(dst, 0, *out_size);
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break;
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}
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} else {
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std::memset(dst, 0, *out_size);
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default:
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// Unknown format - already memset to 0
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LOG_ERROR(Lib_Audio3d, "Unsupported audio format: {}",
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magic_enum::enum_name(pcm.format));
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break;
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}
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} catch (...) {
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// Catch any exceptions (unlikely but safe)
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std::free(dst);
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*out_data = nullptr;
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*out_size = 0;
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}
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}
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static s32 PortQueueAudio(Port& port, const OrbisAudio3dPcm& pcm, const u32 num_channels) {
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u8* dst_data = nullptr;
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u32 dst_size = 0;
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