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Aegisub
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Move the ConvertAudioProvider into GetInt16MonoAudio function 

And use a dummy ConvertAudioProvider to keep backward compatibility.
This commit is contained in:
wangqr 2019-10-29 23:49:55 -04:00
parent 24d52bb1ee
commit c1a4e0674b
4 changed files with 110 additions and 122 deletions
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96
libaegisub/audio/provider.cpp
View File

@ -21,7 +21,103 @@
#include "libaegisub/log.h"
#include "libaegisub/util.h"
namespace {
template<typename Source>
class ConvertFloatToInt16 {
Source* src;
public:
ConvertFloatToInt16(Source* src) :src(src) {}
int16_t operator[](size_t idx) const {
Source expanded = src[idx] * 32768;
return expanded < -32768 ? -32768 :
expanded > 32767 ? 32767 :
static_cast<int16_t>(expanded);
}
};
// 8 bits per sample is assumed to be unsigned with a bias of 128,
// while everything else is assumed to be signed with zero bias
class ConvertIntToInt16 {
void* src;
int bytes_per_sample;
public:
ConvertIntToInt16(void* src, int bytes_per_sample) :src(src), bytes_per_sample(bytes_per_sample) {}
const int16_t& operator[](size_t idx) const {
return *reinterpret_cast<int16_t*>(reinterpret_cast<char*>(src) + (idx + 1) * bytes_per_sample - sizeof(int16_t));
}
};
class ConvertUInt8ToInt16 {
uint8_t* src;
public:
ConvertUInt8ToInt16(uint8_t* src) :src(src) {}
int16_t operator[](size_t idx) const {
return int16_t(src[idx]-128) << 8;
}
};
template<typename Source>
class DownmixToMono {
Source src;
int channels;
public:
DownmixToMono(Source src, int channels) :src(src), channels(channels) {}
int16_t operator[](size_t idx) const {
int ret = 0;
// Just average the channels together
for (int i = 0; i < channels; ++i)
ret += src[idx * channels + i];
return ret / channels;
}
};
}
namespace agi {
void AudioProvider::FillBufferInt16Mono(int16_t* buf, int64_t start, int64_t count) const {
if (!float_samples && bytes_per_sample == 2 && channels == 1) {
FillBuffer(buf, start, count);
return;
}
void* buff = malloc(bytes_per_sample * count * channels);
FillBuffer(buff, start, count);
if (channels == 1) {
if (float_samples) {
if (bytes_per_sample == sizeof(float))
for (int64_t i = 0; i < count; ++i)
buf[i] = ConvertFloatToInt16<float>(reinterpret_cast<float*>(buff))[i];
else if (bytes_per_sample == sizeof(double))
for (int64_t i = 0; i < count; ++i)
buf[i] = ConvertFloatToInt16<double>(reinterpret_cast<double*>(buff))[i];
}
else {
if (bytes_per_sample == sizeof(uint8_t))
for (int64_t i = 0; i < count; ++i)
buf[i] = ConvertUInt8ToInt16(reinterpret_cast<uint8_t*>(buff))[i];
else
for (int64_t i = 0; i < count; ++i)
buf[i] = ConvertIntToInt16(buff, bytes_per_sample)[i];
}
}
else {
if (float_samples) {
if (bytes_per_sample == sizeof(float))
for (int64_t i = 0; i < count; ++i)
buf[i] = DownmixToMono<ConvertFloatToInt16<float> >(ConvertFloatToInt16<float>(reinterpret_cast<float*>(buff)), channels)[i];
else if (bytes_per_sample == sizeof(double))
for (int64_t i = 0; i < count; ++i)
buf[i] = DownmixToMono<ConvertFloatToInt16<double> >(ConvertFloatToInt16<double>(reinterpret_cast<double*>(buff)), channels)[i];
}
else {
if (bytes_per_sample == sizeof(uint8_t))
for (int64_t i = 0; i < count; ++i)
buf[i] = DownmixToMono<ConvertUInt8ToInt16>(ConvertUInt8ToInt16(reinterpret_cast<uint8_t*>(buff)), channels)[i];
else
for (int64_t i = 0; i < count; ++i)
buf[i] = DownmixToMono<ConvertIntToInt16>(ConvertIntToInt16(buff, bytes_per_sample), channels)[i];
}
}
}
void AudioProvider::GetInt16MonoAudioWithVolume(int16_t *buf, int64_t start, int64_t count, double volume) const {
GetInt16MonoAudio(buf, start, count);
if (volume == 1.0) return;

128
libaegisub/audio/provider_convert.cpp
View File

@ -23,115 +23,18 @@
using namespace agi;
/// Anything integral -> 16 bit signed machine-endian audio converter
/// Anything -> mono 16 bit signed machine-endian audio converter
namespace {
template<class Target>
class BitdepthConvertAudioProvider final : public AudioProviderWrapper {
int src_bytes_per_sample;
mutable std::vector<uint8_t> src_buf;
class ConvertAudioProvider final : public AudioProviderWrapper {
public:
BitdepthConvertAudioProvider(std::unique_ptr<AudioProvider> src) : AudioProviderWrapper(std::move(src)) {
if (bytes_per_sample > 8)
throw AudioProviderError("Audio format converter: audio with bitdepths greater than 64 bits/sample is currently unsupported");
src_bytes_per_sample = bytes_per_sample;
bytes_per_sample = sizeof(Target);
}
void FillBuffer(void *buf, int64_t start, int64_t count64) const override {
auto count = static_cast<size_t>(count64);
assert(count64 >= 0 && count == static_cast<uint64_t>(count64));
src_buf.resize(count * src_bytes_per_sample * channels);
source->GetAudio(src_buf.data(), start, count);
auto dest = static_cast<int16_t*>(buf);
for (size_t i = 0; i < count * channels; ++i) {
int64_t sample = 0;
// 8 bits per sample is assumed to be unsigned with a bias of 127,
// while everything else is assumed to be signed with zero bias
if (src_bytes_per_sample == 1)
sample = src_buf[i] - 128;
else {
for (int j = src_bytes_per_sample; j > 0; --j) {
sample <<= 8;
sample += src_buf[i * src_bytes_per_sample + j - 1];
}
}
if (static_cast<size_t>(src_bytes_per_sample) > sizeof(Target))
sample /= 1LL << (src_bytes_per_sample - sizeof(Target)) * 8;
else if (static_cast<size_t>(src_bytes_per_sample) < sizeof(Target))
sample *= 1LL << (sizeof(Target) - src_bytes_per_sample ) * 8;
dest[i] = static_cast<Target>(sample);
}
}
};
/// 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);
ConvertAudioProvider(std::unique_ptr<AudioProvider> src) : AudioProviderWrapper(std::move(src)) {
float_samples = false;
}
void FillBuffer(void *buf, int64_t start, int64_t count64) const override {
auto count = static_cast<size_t>(count64);
assert(count64 >= 0 && count == static_cast<uint64_t>(count64));
src_buf.resize(count * channels);
source->GetAudio(&src_buf[0], start, count);
auto dest = static_cast<Target*>(buf);
for (size_t i = 0; i < static_cast<size_t>(count * channels); ++i) {
Source expanded;
if (src_buf[i] < 0)
expanded = static_cast<Target>(-src_buf[i] * std::numeric_limits<Target>::min());
else
expanded = static_cast<Target>(src_buf[i] * std::numeric_limits<Target>::max());
dest[i] = expanded < std::numeric_limits<Target>::min() ? std::numeric_limits<Target>::min() :
expanded > std::numeric_limits<Target>::max() ? std::numeric_limits<Target>::max() :
static_cast<Target>(expanded);
}
}
};
/// 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)) {
src_channels = channels;
channels = 1;
bytes_per_sample = sizeof(int16_t);
}
void FillBuffer(void *buf, int64_t start, int64_t count64) const override {
auto count = static_cast<size_t>(count64);
assert(count64 >= 0 && count == static_cast<uint64_t>(count64));
src_buf.resize(count * src_channels);
source->GetAudio(&src_buf[0], start, count);
auto dst = static_cast<int16_t*>(buf);
// Just average the channels together
while (count-- > 0) {
int sum = 0;
for (int c = 0; c < src_channels; ++c)
sum += src_buf[count * src_channels + c];
dst[count] = static_cast<int16_t>(sum / src_channels);
}
void FillBuffer(void *buf, int64_t start, int64_t count) const override {
source->GetInt16MonoAudio(reinterpret_cast<int16_t*>(buf), start, count);
}
};
@ -175,23 +78,16 @@ public:
namespace agi {
std::unique_ptr<AudioProvider> CreateConvertAudioProvider(std::unique_ptr<AudioProvider> provider) {
// Ensure 16-bit audio with proper endianness
if (provider->AreSamplesFloat()) {
if (provider->AreSamplesFloat())
LOG_D("audio_provider") << "Converting float to S16";
if (provider->GetBytesPerSample() == sizeof(float))
provider = agi::make_unique<FloatConvertAudioProvider<float, int16_t>>(std::move(provider));
else
provider = agi::make_unique<FloatConvertAudioProvider<double, int16_t>>(std::move(provider));
}
if (provider->GetBytesPerSample() != 2) {
LOG_D("audio_provider") << "Converting " << provider->GetBytesPerSample() << " bytes per sample or wrong endian to S16";
provider = agi::make_unique<BitdepthConvertAudioProvider<int16_t>>(std::move(provider));
}
else if (provider->GetBytesPerSample() != 2)
LOG_D("audio_provider") << "Converting " << provider->GetBytesPerSample() << " bytes per sample to S16";
// We currently only support mono audio
if (provider->GetChannels() != 1) {
if (provider->GetChannels() != 1)
LOG_D("audio_provider") << "Downmixing to mono from " << provider->GetChannels() << " channels";
provider = agi::make_unique<DownmixAudioProvider>(std::move(provider));
}
provider = agi::make_unique<ConvertAudioProvider>(std::move(provider));
// Some players don't like low sample rate audio
while (provider->GetSampleRate() < 32000) {

6
libaegisub/include/libaegisub/audio/provider.h
View File

@ -36,11 +36,7 @@ protected:
bool float_samples = false;
virtual void FillBuffer(void *buf, int64_t start, int64_t count) const = 0;
virtual void FillBufferInt16Mono(int16_t* buf, int64_t start, int64_t count) const {
if (float_samples || bytes_per_sample != 2 || channels != 1)
throw agi::InternalError("FillBufferInt16Mono called on unconverted audio stream");
FillBuffer(buf, start, count);
}
virtual void FillBufferInt16Mono(int16_t* buf, int64_t start, int64_t count) const;
void ZeroFill(void *buf, int64_t count) const;

2
tests/tests/audio.cpp
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@ -333,7 +333,7 @@ struct FloatAudioProvider : agi::AudioProvider {
auto out = static_cast<Float *>(buf);
for (int64_t end = start + count; start < end; ++start) {
auto shifted = start + SHRT_MIN;
*out++ = (Float)(1.0 * shifted / (shifted < 0 ? -SHRT_MIN : SHRT_MAX));
*out++ = (Float)(shifted) / (-SHRT_MIN);
}
}
};