Aegisub/libaegisub/audio/provider_convert.cpp
wangqr c2c44f1ad2 Fix build warnings
For pimpl with anonymous namespace, see https://stackoverflow.com/questions/39684438
2019-09-07 01:31:16 -04:00

204 lines
7.1 KiB
C++

// Copyright (c) 2014, Thomas Goyne <plorkyeran@aegisub.org>
//
// Permission to use, copy, modify, and distribute this software for any
// purpose with or without fee is hereby granted, provided that the above
// copyright notice and this permission notice appear in all copies.
//
// THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
// WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
// MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
// ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
// WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
// ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
// OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
//
// Aegisub Project http://www.aegisub.org/
#include "libaegisub/audio/provider.h"
#include <libaegisub/log.h>
#include <libaegisub/make_unique.h>
#include <limits>
using namespace agi;
/// Anything integral -> 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;
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);
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;
}
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);
}
}
};
/// 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)) {
sample_rate *= 2;
num_samples *= 2;
decoded_samples = decoded_samples * 2;
}
void FillBuffer(void *buf, int64_t start, int64_t count) const override {
int16_t *src, *dst = static_cast<int16_t *>(buf);
// We need to always get at least two samples to be able to interpolate
int16_t srcbuf[2];
if (count == 1) {
source->GetAudio(srcbuf, start / 2, 2);
src = srcbuf;
}
else {
source->GetAudio(buf, start / 2, (start + count) / 2 - start / 2 + 1);
src = dst;
}
// walking backwards so that the conversion can be done in place
for (; count > 0; --count) {
auto src_index = (start + count - 1) / 2 - start / 2;
auto i = count - 1;
if ((start + i) & 1)
dst[i] = (int16_t)(((int32_t)src[src_index] + src[src_index + 1]) / 2);
else
dst[i] = src[src_index];
}
}
};
}
namespace agi {
std::unique_ptr<AudioProvider> CreateConvertAudioProvider(std::unique_ptr<AudioProvider> provider) {
// Ensure 16-bit audio with proper endianness
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));
}
// We currently only support mono audio
if (provider->GetChannels() != 1) {
LOG_D("audio_provider") << "Downmixing to mono from " << provider->GetChannels() << " channels";
provider = agi::make_unique<DownmixAudioProvider>(std::move(provider));
}
// Some players don't like low sample rate audio
while (provider->GetSampleRate() < 32000) {
LOG_D("audio_provider") << "Doubling sample rate";
provider = agi::make_unique<SampleDoublingAudioProvider>(std::move(provider));
}
return provider;
}
}