Aegisub/libaegisub/audio/provider.cpp

306 lines
10 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/fs.h"
#include "libaegisub/io.h"
#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];
}
}
free(buff);
}
// This entire file has turned into a mess. For now I'm just following the pattern of the wangqr code, but
// this should really be restructured entirely again. The original type constructor-based system worked very well - it could
// just give downmix/conversion control to the players instead.
void AudioProvider::GetAudioWithVolume(void *buf, int64_t start, int64_t count, double volume) const {
GetAudio(buf, start, count);
if (volume == 1.0) return;
int64_t n = count * GetChannels();
if (float_samples) {
if (bytes_per_sample == sizeof(float)) {
float *buff = reinterpret_cast<float *>(buf);
for (int64_t i = 0; i < n; ++i)
buff[i] = static_cast<float>(buff[i] * volume);
} else if (bytes_per_sample == sizeof(double)) {
double *buff = reinterpret_cast<double *>(buf);
for (int64_t i = 0; i < n; ++i)
buff[i] = buff[i] * volume;
}
}
else {
if (bytes_per_sample == sizeof(uint8_t)) {
uint8_t *buff = reinterpret_cast<uint8_t *>(buf);
for (int64_t i = 0; i < n; ++i)
buff[i] = util::mid(0, static_cast<int>(((int) buff[i] - 128) * volume + 128), 0xFF);
} else if (bytes_per_sample == sizeof(int16_t)) {
int16_t *buff = reinterpret_cast<int16_t *>(buf);
for (int64_t i = 0; i < n; ++i)
buff[i] = util::mid(-0x8000, static_cast<int>(buff[i] * volume), 0x7FFF);
} else if (bytes_per_sample == sizeof(int32_t)) {
int32_t *buff = reinterpret_cast<int32_t *>(buf);
for (int64_t i = 0; i < n; ++i)
buff[i] = static_cast<int32_t>(buff[i] * volume);
} else if (bytes_per_sample == sizeof(int64_t)) {
int64_t *buff = reinterpret_cast<int64_t *>(buf);
for (int64_t i = 0; i < n; ++i)
buff[i] = static_cast<int64_t>(buff[i] * volume);
}
}
}
void AudioProvider::GetInt16MonoAudioWithVolume(int16_t *buf, int64_t start, int64_t count, double volume) const {
GetInt16MonoAudio(buf, start, count);
if (volume == 1.0) return;
auto buffer = static_cast<int16_t *>(buf);
for (size_t i = 0; i < (size_t)count; ++i)
buffer[i] = util::mid(-0x8000, static_cast<int>(buffer[i] * volume + 0.5), 0x7FFF);
}
void AudioProvider::ZeroFill(void *buf, int64_t count) const {
if (bytes_per_sample == 1)
// 8 bit formats are usually unsigned with bias 128
memset(buf, 128, count * channels);
else // While everything else is signed
memset(buf, 0, count * bytes_per_sample * channels);
}
void AudioProvider::GetAudio(void *buf, int64_t start, int64_t count) const {
if (start < 0) {
ZeroFill(buf, std::min(-start, count));
buf = static_cast<char *>(buf) + -start * bytes_per_sample * channels;
count += start;
start = 0;
}
if (start + count > num_samples) {
int64_t zero_count = std::min(count, start + count - num_samples);
count -= zero_count;
ZeroFill(static_cast<char *>(buf) + count * bytes_per_sample * channels, zero_count);
}
if (count <= 0) return;
try {
FillBuffer(buf, start, count);
}
catch (AudioDecodeError const& e) {
// We don't have any good way to report errors here, so just log the
// failure and return silence
LOG_E("audio_provider") << e.GetMessage();
ZeroFill(buf, count);
return;
}
catch (...) {
LOG_E("audio_provider") << "Unknown audio decoding error";
ZeroFill(buf, count);
return;
}
}
void AudioProvider::GetInt16MonoAudio(int16_t* buf, int64_t start, int64_t count) const {
if (start < 0) {
memset(buf, 0, sizeof(int16_t) * std::min(-start, count));
buf -= start;
count += start;
start = 0;
}
if (start + count > num_samples) {
int64_t zero_count = std::min(count, start + count - num_samples);
count -= zero_count;
memset(buf + count, 0, sizeof(int16_t) * zero_count);
}
if (count <= 0) return;
try {
FillBufferInt16Mono(buf, start, count);
}
catch (AudioDecodeError const& e) {
// We don't have any good way to report errors here, so just log the
// failure and return silence
LOG_E("audio_provider") << e.GetMessage();
memset(buf, 0, sizeof(int16_t) * count);
return;
}
catch (...) {
LOG_E("audio_provider") << "Unknown audio decoding error";
memset(buf, 0, sizeof(int16_t) * count);
return;
}
}
namespace {
class writer {
io::Save outfile;
std::ostream& out;
public:
writer(agi::fs::path const& filename) : outfile(filename, true), out(outfile.Get()) { }
template<int N>
void write(const char(&str)[N]) {
out.write(str, N - 1);
}
void write(std::vector<char> const& data) {
out.write(data.data(), data.size());
}
template<typename Dest, typename Src>
void write(Src v) {
auto converted = static_cast<Dest>(v);
out.write(reinterpret_cast<char *>(&converted), sizeof(Dest));
}
};
}
void SaveAudioClip(AudioProvider const& provider, fs::path const& path, int start_time, int end_time) {
const auto max_samples = provider.GetNumSamples();
const auto start_sample = std::min(max_samples, ((int64_t)start_time * provider.GetSampleRate() + 999) / 1000);
const auto end_sample = util::mid(start_sample, ((int64_t)end_time * provider.GetSampleRate() + 999) / 1000, max_samples);
const size_t bytes_per_sample = provider.GetBytesPerSample() * provider.GetChannels();
const size_t bufsize = (end_sample - start_sample) * bytes_per_sample;
writer out{path};
out.write("RIFF");
out.write<int32_t>(bufsize + 36);
out.write("WAVEfmt ");
out.write<int32_t>(16); // Size of chunk
out.write<int16_t>(provider.AreSamplesFloat() ? 3 : 1); // compression format (1: WAVE_FORMAT_PCM, 3: WAVE_FORMAT_IEEE_FLOAT)
out.write<int16_t>(provider.GetChannels());
out.write<int32_t>(provider.GetSampleRate());
out.write<int32_t>(provider.GetSampleRate() * provider.GetChannels() * provider.GetBytesPerSample());
out.write<int16_t>(provider.GetChannels() * provider.GetBytesPerSample());
out.write<int16_t>(provider.GetBytesPerSample() * 8);
out.write("data");
out.write<int32_t>(bufsize);
// samples per read
size_t spr = 65536 / bytes_per_sample;
std::vector<char> buf;
for (int64_t i = start_sample; i < end_sample; i += spr) {
spr = std::min<size_t>(spr, end_sample - i);
buf.resize(spr * bytes_per_sample);
provider.GetAudio(&buf[0], i, spr);
out.write(buf);
}
}
}