forked from mia/Aegisub
0336779735
Removed downsampling in FFMS2 and CreateConvertAudioProvider, to ensure we can get the original audio channels and data. Fix Aegisub/Aegisub#160
101 lines
3.5 KiB
C++
101 lines
3.5 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 -> mono 16 bit signed machine-endian audio converter
|
|
namespace {
|
|
class ConvertAudioProvider final : public AudioProviderWrapper {
|
|
public:
|
|
ConvertAudioProvider(std::unique_ptr<AudioProvider> src) : AudioProviderWrapper(std::move(src)) {
|
|
float_samples = false;
|
|
channels = 1;
|
|
bytes_per_sample = sizeof(int16_t);
|
|
}
|
|
|
|
void FillBuffer(void *buf, int64_t start, int64_t count) const override {
|
|
source->GetInt16MonoAudio(reinterpret_cast<int16_t*>(buf), start, count);
|
|
}
|
|
};
|
|
|
|
/// 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";
|
|
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)
|
|
LOG_D("audio_provider") << "Downmixing to mono from " << provider->GetChannels() << " channels";
|
|
|
|
// Some players don't like low sample rate audio
|
|
if (provider->GetSampleRate() < 32000) {
|
|
provider = agi::make_unique<ConvertAudioProvider>(std::move(provider));
|
|
while (provider->GetSampleRate() < 32000) {
|
|
LOG_D("audio_provider") << "Doubling sample rate";
|
|
provider = agi::make_unique<SampleDoublingAudioProvider>(std::move(provider));
|
|
}
|
|
}
|
|
|
|
return provider;
|
|
}
|
|
}
|