Aegisub/aegisub/libmedia/audio/convert.cpp
Amar Takhar f16fb6bb5f Add fix convert and downmix audio providers.
Originally committed to SVN as r5302.
2011-02-06 03:09:59 +00:00

225 lines
6.3 KiB
C++

// Copyright (c) 2008, Rodrigo Braz Monteiro
// All rights reserved.
//
// Redistribution and use in source and binary forms, with or without
// modification, are permitted provided that the following conditions are met:
//
// * Redistributions of source code must retain the above copyright notice,
// this list of conditions and the following disclaimer.
// * Redistributions in binary form must reproduce the above copyright notice,
// this list of conditions and the following disclaimer in the documentation
// and/or other materials provided with the distribution.
// * Neither the name of the Aegisub Group nor the names of its contributors
// may be used to endorse or promote products derived from this software
// without specific prior written permission.
//
// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
// AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
// IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
// ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
// LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
// CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
// SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
// INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
// CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
// ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
// POSSIBILITY OF SUCH DAMAGE.
//
// Aegisub Project http://www.aegisub.org/
//
// $Id$
/// @file audio_provider_convert.cpp
/// @brief Intermediate sample format-converting audio provider
/// @ingroup audio_input
///
#include "config.h"
#include "aegisub_endian.h"
#include "convert.h"
#include "downmix.h"
namespace media {
/// @brief Constructor
/// @param src
///
ConvertAudioProvider::ConvertAudioProvider(AudioProvider *src) : source(src) {
channels = source->GetChannels();
num_samples = source->GetNumSamples();
sample_rate = source->GetSampleRate();
bytes_per_sample = 2;
sampleMult = 1;
if (sample_rate < 16000) sampleMult = 4;
else if (sample_rate < 32000) sampleMult = 2;
sample_rate *= sampleMult;
num_samples *= sampleMult;
}
/// @brief Convert to 16-bit
/// @param src
/// @param dst
/// @param count
///
void ConvertAudioProvider::Make16Bit(const char *src, short *dst, int64_t count) const {
for (int64_t i=0;i<count;i++) {
dst[i] = (short(src[i])-128)*255;
}
}
//////////////////////
// Change sample rate
// This requres 16-bit input
// The SampleConverter is a class overloading operator() with a function from short to short
template<class SampleConverter>
/// @brief DOCME
/// @param src
/// @param dst
/// @param count
/// @param converter
///
void ConvertAudioProvider::ChangeSampleRate(const short *src, short *dst, int64_t count, const SampleConverter &converter) const {
// Upsample by 2
if (sampleMult == 2) {
int64_t size = count/2;
short cur;
short next = 0;
for (int64_t i=0;i<size;i++) {
cur = next;
next = converter(*src++);
*(dst++) = cur;
*(dst++) = (cur+next)/2;
}
if (count%2) *(dst++) = next;
}
// Upsample by 4
else if (sampleMult == 4) {
int64_t size = count/4;
short cur;
short next = 0;
for (int64_t i=0;i<size;i++) {
cur = next;
next = converter(*src++);
*(dst++) = cur;
*(dst++) = (cur*3+next)/4;
*(dst++) = (cur+next)/2;
*(dst++) = (cur+next*3)/4;
}
for (int i=0;i<count%4;i++) *(dst++) = next;
}
// Nothing much to do, just ensure correct endedness
else if (sampleMult == 1) {
while (count-- > 0) {
*dst++ = converter(*src++);
}
}
}
/// DOCME
struct NullSampleConverter {
inline short operator()(const short val) const {
return val;
}
};
/// DOCME
struct EndianSwapSampleConverter {
inline short operator()(const short val) const {
return (short)Endian::Reverse((uint16_t)val);
};
};
/// @brief Get audio
/// @param destination
/// @param start
/// @param count
///
void ConvertAudioProvider::GetAudio(void *destination, int64_t start, int64_t count) const {
// Bits per sample
int srcBps = source->GetBytesPerSample();
// Nothing to do
if (sampleMult == 1 && srcBps == 2) {
source->GetAudio(destination,start,count);
}
// Convert
else {
// Allocate buffers with sufficient size for the entire operation
size_t fullSize = count;
int64_t srcCount = count / sampleMult;
short *buffer1 = NULL;
short *buffer2 = NULL;
short *last = NULL;
// Read audio
buffer1 = new short[fullSize * channels];
source->GetAudio(buffer1,start/sampleMult,srcCount);
// Convert from 8-bit to 16-bit
if (srcBps == 1) {
if (sampleMult == 1) {
Make16Bit((const char*)buffer1,(short*)destination,srcCount * channels);
}
else {
buffer2 = new short[fullSize * channels];
Make16Bit((const char*)buffer1,buffer2,srcCount * channels);
last = buffer2;
}
}
// Already 16-bit
else if (srcBps == 2) last = buffer1;
// Convert sample rate
if (sampleMult != 1 && source->AreSamplesNativeEndian()) {
ChangeSampleRate(last,(short*)destination,count * channels, NullSampleConverter());
}
else if (!source->AreSamplesNativeEndian()) {
ChangeSampleRate(last,(short*)destination,count * channels, EndianSwapSampleConverter());
}
delete [] buffer1;
delete [] buffer2;
}
}
/// @brief See if we need to downmix the number of channels
/// @param source_provider
///
AudioProvider *CreateConvertAudioProvider(AudioProvider *source_provider) {
AudioProvider *provider = source_provider;
// Aegisub requires 16 bit samples,
// some audio players break with low samplerates,
// everything breaks with wrong-ended samples.
if (provider->GetBytesPerSample() != 2 ||
provider->GetSampleRate() < 32000 ||
!provider->AreSamplesNativeEndian())
{
// @todo add support for more bitdepths (i.e. 24- and 32-bit audio)
if (provider->GetBytesPerSample() > 2)
throw AudioOpenError("Audio format converter: audio with bitdepths greater than 16 bits/sample is currently unsupported");
provider = new ConvertAudioProvider(provider);
}
// We also require mono audio for historical reasons
if (provider->GetChannels() != 1)
{
provider = new DownmixingAudioProvider(provider);
}
return provider;
}
} // namespace