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Aegisub
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Reuse the decoding buffer in the audio converters

This commit is contained in:
Thomas Goyne 2014-07-10 08:11:59 -07:00
parent b9c75d8706
commit 93ad15efb1
1 changed files with 12 additions and 19 deletions
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31
libaegisub/audio/provider_convert.cpp
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@ -28,6 +28,8 @@ namespace {
template<class Target>
class BitdepthConvertAudioProvider final : public AudioProviderWrapper {
int src_bytes_per_sample;
mutable std::vector<uint8_t> src_buf;
public:
BitdepthConvertAudioProvider(std::unique_ptr<AudioProvider> src) : AudioProviderWrapper(std::move(src)) {
if (bytes_per_sample > 8)
@ -38,7 +40,7 @@ public:
}
void FillBuffer(void *buf, int64_t start, int64_t count) const override {
std::vector<uint8_t> src_buf(count * src_bytes_per_sample * channels);
src_buf.resize(count * src_bytes_per_sample * channels);
source->GetAudio(src_buf.data(), start, count);
auto dest = static_cast<int16_t*>(buf);
@ -70,6 +72,8 @@ public:
/// Floating point -> 16 bit signed machine-endian audio converter
template<class Source, class Target>
class FloatConvertAudioProvider final : public AudioProviderWrapper {
mutable std::vector<Source> src_buf;
public:
FloatConvertAudioProvider(std::unique_ptr<AudioProvider> src) : AudioProviderWrapper(std::move(src)) {
bytes_per_sample = sizeof(Target);
@ -77,10 +81,10 @@ public:
}
void FillBuffer(void *buf, int64_t start, int64_t count) const override {
std::vector<Source> src_buf(count * channels);
src_buf.resize(count * channels);
source->GetAudio(&src_buf[0], start, count);
auto dest = reinterpret_cast<Target*>(buf);
auto dest = static_cast<Target*>(buf);
for (size_t i = 0; i < static_cast<size_t>(count * channels); ++i) {
Source expanded;
@ -102,23 +106,19 @@ public:
/// Non-mono 16-bit signed machine-endian -> mono 16-bit signed machine endian converter
class DownmixAudioProvider final : public AudioProviderWrapper {
int src_channels;
mutable std::vector<int16_t> src_buf;
public:
DownmixAudioProvider(std::unique_ptr<AudioProvider> src) : AudioProviderWrapper(std::move(src)) {
if (bytes_per_sample != 2)
throw InternalError("DownmixAudioProvider requires 16-bit input");
if (channels == 1)
throw InternalError("DownmixAudioProvider requires multi-channel input");
src_channels = channels;
channels = 1;
}
void FillBuffer(void *buf, int64_t start, int64_t count) const override {
if (count == 0) return;
std::vector<int16_t> src_buf(count * src_channels);
src_buf.resize(count * src_channels);
source->GetAudio(&src_buf[0], start, count);
int16_t *dst = reinterpret_cast<int16_t*>(buf);
auto dst = static_cast<int16_t*>(buf);
// Just average the channels together
while (count-- > 0) {
int sum = 0;
@ -129,24 +129,17 @@ public:
}
};
/// Sample doubler with linear interpolation for the agi::make_unique<samples>
/// Sample doubler with linear interpolation for the samples provider
/// Requires 16-bit mono input
class SampleDoublingAudioProvider final : public AudioProviderWrapper {
public:
SampleDoublingAudioProvider(std::unique_ptr<AudioProvider> src) : AudioProviderWrapper(std::move(src)) {
if (source->GetBytesPerSample() != 2)
throw InternalError("UpsampleAudioProvider requires 16-bit input");
if (source->GetChannels() != 1)
throw InternalError("UpsampleAudioProvider requires mono input");
sample_rate *= 2;
num_samples *= 2;
decoded_samples = decoded_samples * 2;
}
void FillBuffer(void *buf, int64_t start, int64_t count) const override {
if (count == 0) return;
bool not_end = start + count < num_samples;
int64_t src_count = count / 2;
source->GetAudio(buf, start / 2, src_count + not_end);