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Fix: do not overwrite the output buffer when it is accumulative.

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Fixes https://github.com/AndroidAudioMods/ViPER4Android/issues/16
This commit is contained in:
Iscle 2023-01-25 01:48:00 +01:00
parent a7d5eb60cd
commit 2e9a84134b
1 changed files with 39 additions and 18 deletions
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57
src/ViPER4Android.cpp
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@ -39,19 +39,37 @@ static void pcm32ToFloat(float *dst, const int32_t *src, size_t frameCount) {
}
}
static void floatToFloat(float *dst, const float *src, size_t frameCount) {
memcpy(dst, src, frameCount * 2 * sizeof(float));
}
static void floatToPcm16(int16_t *dst, const float *src, size_t frameCount) {
for (size_t i = 0; i < frameCount * 2; i++) {
dst[i] = (int16_t) round(src[i] * (float) (1 << 15));
static void floatToFloat(float *dst, const float *src, size_t frameCount, bool accumulate) {
if (accumulate) {
for (size_t i = 0; i < frameCount * 2; i++) {
dst[i] += src[i];
}
} else {
memcpy(dst, src, frameCount * 2 * sizeof(float));
}
}
static void floatToPcm32(int32_t *dst, const float *src, size_t frameCount) {
for (size_t i = 0; i < frameCount * 2; i++) {
dst[i] = (int32_t) round(src[i] * (float) (1 << 31));
static void floatToPcm16(int16_t *dst, const float *src, size_t frameCount, bool accumulate) {
if (accumulate) {
for (size_t i = 0; i < frameCount * 2; i++) {
dst[i] += (int16_t) round(src[i] * (float) (1 << 15));
}
} else {
for (size_t i = 0; i < frameCount * 2; i++) {
dst[i] = (int16_t) round(src[i] * (float) (1 << 15));
}
}
}
static void floatToPcm32(int32_t *dst, const float *src, size_t frameCount, bool accumulate) {
if (accumulate) {
for (size_t i = 0; i < frameCount * 2; i++) {
dst[i] += (int32_t) round(src[i] * (float) (1 << 31));
}
} else {
for (size_t i = 0; i < frameCount * 2; i++) {
dst[i] = (int32_t) round(src[i] * (float) (1 << 31));
}
}
}
@ -66,41 +84,42 @@ static int32_t Viper_IProcess(effect_handle_t self, audio_buffer_t *inBuffer, au
return -EINVAL;
}
float *buffer;
float *buffer = new float[outBuffer->frameCount * 2];
size_t frameCount = outBuffer->frameCount;
switch (pContext->config.inputCfg.format) {
case AUDIO_FORMAT_PCM_16_BIT:
buffer = new float[outBuffer->frameCount * 2];
pcm16ToFloat(buffer, inBuffer->s16, frameCount);
break;
case AUDIO_FORMAT_PCM_32_BIT:
buffer = new float[outBuffer->frameCount * 2];
pcm32ToFloat(buffer, inBuffer->s32, frameCount);
break;
case AUDIO_FORMAT_PCM_FLOAT:
buffer = outBuffer->f32;
floatToFloat(buffer, inBuffer->f32, frameCount);
floatToFloat(buffer, inBuffer->f32, frameCount, false);
break;
default:
delete[] buffer;
return -EINVAL;
}
pContext->viper->processBuffer(buffer, frameCount);
const bool accumulate = pContext->config.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE;
switch (pContext->config.outputCfg.format) {
case AUDIO_FORMAT_PCM_16_BIT:
floatToPcm16(outBuffer->s16, buffer, frameCount);
floatToPcm16(outBuffer->s16, buffer, frameCount, accumulate);
delete[] buffer;
break;
case AUDIO_FORMAT_PCM_32_BIT:
floatToPcm32(outBuffer->s32, buffer, frameCount);
floatToPcm32(outBuffer->s32, buffer, frameCount, accumulate);
delete[] buffer;
break;
case AUDIO_FORMAT_PCM_FLOAT:
floatToFloat(outBuffer->f32, buffer, frameCount);
floatToFloat(outBuffer->f32, buffer, frameCount, accumulate);
delete[] buffer;
break;
default:
delete[] buffer;
return -EINVAL;
}
@ -114,9 +133,11 @@ static int handleSetConfig(ViperContext *pContext, effect_config_t *newConfig) {
VIPER_LOGI("Input sampling rate: %d", newConfig->inputCfg.samplingRate);
VIPER_LOGI("Input channels: %d", newConfig->inputCfg.channels);
VIPER_LOGI("Input format: %d", newConfig->inputCfg.format);
VIPER_LOGI("Input access mode: %d", newConfig->inputCfg.accessMode);
VIPER_LOGI("Output sampling rate: %d", newConfig->outputCfg.samplingRate);
VIPER_LOGI("Output channels: %d", newConfig->outputCfg.channels);
VIPER_LOGI("Output format: %d", newConfig->outputCfg.format);
VIPER_LOGI("Output access mode: %d", newConfig->outputCfg.accessMode);
pContext->isConfigValid = false;