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https://github.com/AndroidAudioMods/ViPERFX_RE.git
synced 2024-12-22 22:47:25 +08:00
Improve buffer processing and pcm conversion
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parent
50f1bd350a
commit
3f114fdbb4
@ -362,55 +362,51 @@ int32_t ViperContext::handleCommand(uint32_t cmdCode, uint32_t cmdSize, void *pC
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}
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}
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static void pcm16ToFloat(float *dst, const int16_t *src, size_t frameCount) {
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template <typename T>
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void pcmToFloat(float* dst, const T* src, size_t frameCount) {
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constexpr float max_val = static_cast<float>(std::numeric_limits<T>::max());
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for (size_t i = 0; i < frameCount * 2; i++) {
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dst[i] = static_cast<float>(src[i]) / static_cast<float>(1 << 15);
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dst[i] = static_cast<float>(src[i]) / max_val;
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}
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}
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static void pcm32ToFloat(float *dst, const int32_t *src, size_t frameCount) {
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for (size_t i = 0; i < frameCount * 2; i++) {
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dst[i] = static_cast<float>(src[i]) / static_cast<float>(1 << 31);
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}
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template <typename T>
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static const T& clamp(const T& v, const T& lo, const T& hi) {
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return std::min(std::max(v, lo), hi);
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}
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static void floatToFloat(float *dst, const float *src, size_t frameCount, bool accumulate) {
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if (accumulate) {
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for (size_t i = 0; i < frameCount * 2; i++) {
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dst[i] += src[i];
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dst[i] += clamp(src[i], -1.0f, 1.0f);
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}
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} else {
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memcpy(dst, src, frameCount * 2 * sizeof(float));
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}
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}
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static void floatToPcm16(int16_t *dst, const float *src, size_t frameCount, bool accumulate) {
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if (accumulate) {
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for (size_t i = 0; i < frameCount * 2; i++) {
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dst[i] += static_cast<int16_t>(std::roundf(src[i] * static_cast<float>(1 << 15)));
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}
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} else {
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for (size_t i = 0; i < frameCount * 2; i++) {
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dst[i] = static_cast<int16_t>(std::roundf(src[i] * static_cast<float>(1 << 15)));
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}
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}
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}
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template <typename T, typename U>
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void floatToPcm(T *dst, const float *src, size_t frameCount, bool accumulate) {
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constexpr T max_val = std::numeric_limits<T>::max();
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constexpr T min_val = std::numeric_limits<T>::min();
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static void floatToPcm32(int32_t *dst, const float *src, size_t frameCount, bool accumulate) {
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for (size_t i = 0; i < frameCount * 2; i++) {
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float f = clamp(src[i], -1.0f, 1.0f);
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T pcm = static_cast<T>(f * static_cast<float>(max_val));
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if (accumulate) {
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for (size_t i = 0; i < frameCount * 2; i++) {
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dst[i] += static_cast<int32_t>(std::roundf(src[i] * static_cast<float>(1 << 31)));
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}
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U temp = static_cast<U>(dst[i]) + pcm;
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dst[i] = static_cast<T>(clamp(temp, static_cast<U>(min_val), static_cast<U>(max_val)));
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} else {
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for (size_t i = 0; i < frameCount * 2; i++) {
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dst[i] = static_cast<int32_t>(std::roundf(src[i] * static_cast<float>(1 << 31)));
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dst[i] = pcm;
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}
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}
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}
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static audio_buffer_t *getBuffer(buffer_config_s *config, audio_buffer_t *buffer) {
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if (buffer != nullptr) return buffer;
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return &config->buffer;
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if (config->mask & EFFECT_CONFIG_BUFFER) return &config->buffer;
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// EFFECT_CONFIG_PROVIDER not implemented, it's not used by any known effect
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return nullptr;
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}
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int32_t ViperContext::process(audio_buffer_t *inBuffer, audio_buffer_t *outBuffer) {
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@ -433,16 +429,17 @@ int32_t ViperContext::process(audio_buffer_t *inBuffer, audio_buffer_t *outBuffe
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size_t frameCount = inBuffer->frameCount;
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if (frameCount > bufferFrameCount) {
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// This should never happen, but just in case
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buffer.resize(frameCount * 2);
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bufferFrameCount = frameCount;
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}
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switch (config.inputCfg.format) {
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case AUDIO_FORMAT_PCM_16_BIT:
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pcm16ToFloat(buffer.data(), inBuffer->s16, frameCount);
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pcmToFloat<int16_t>(buffer.data(), inBuffer->s16, frameCount);
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break;
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case AUDIO_FORMAT_PCM_32_BIT:
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pcm32ToFloat(buffer.data(), inBuffer->s32, frameCount);
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pcmToFloat<int32_t>(buffer.data(), inBuffer->s32, frameCount);
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break;
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case AUDIO_FORMAT_PCM_FLOAT:
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floatToFloat(buffer.data(), inBuffer->f32, frameCount, false);
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@ -456,10 +453,10 @@ int32_t ViperContext::process(audio_buffer_t *inBuffer, audio_buffer_t *outBuffe
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const bool accumulate = config.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE;
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switch (config.outputCfg.format) {
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case AUDIO_FORMAT_PCM_16_BIT:
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floatToPcm16(outBuffer->s16, buffer.data(), frameCount, accumulate);
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floatToPcm<int16_t, int32_t>(outBuffer->s16, buffer.data(), frameCount, accumulate);
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break;
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case AUDIO_FORMAT_PCM_32_BIT:
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floatToPcm32(outBuffer->s32, buffer.data(), frameCount, accumulate);
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floatToPcm<int32_t, int64_t>(outBuffer->s32, buffer.data(), frameCount, accumulate);
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break;
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case AUDIO_FORMAT_PCM_FLOAT:
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floatToFloat(outBuffer->f32, buffer.data(), frameCount, accumulate);
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