Aegisub/aegisub/audio_provider_pcm.cpp
Niels Martin Hansen 89d076c760 Apparently long long isn't safe, trying with int64_t instead then...
Originally committed to SVN as r1547.
2007-08-31 14:11:35 +00:00

298 lines
9 KiB
C++

// Copyright (c) 2007, Niels Martin Hansen
// 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
//
// Website: http://aegisub.cellosoft.com
// Contact: mailto:jiifurusu@gmail.com
//
#include <wx/filename.h>
#include <wx/file.h>
#include "audio_provider_pcm.h"
#include "utils.h"
#include <stdint.h>
void PCMAudioProvider::GetAudio(void *buf, int64_t start, int64_t count)
{
// We'll be seeking in the file so state can become inconsistent
wxMutexLocker _fml(filemutex);
// Read blocks from the file
size_t index = 0;
while (count > 0 && index < index_points.size()) {
// Check if this index contains the samples we're looking for
IndexPoint &ip = index_points[index];
if (ip.start_sample <= start && ip.start_sample+ip.num_samples > start) {
// How many samples we can maximum take from this block
int64_t samples_can_do = ip.num_samples - start + ip.start_sample;
if (samples_can_do > count) samples_can_do = count;
// Read as many samples we can
file.Seek(ip.start_byte + (start - ip.start_sample) * bytes_per_sample * channels, wxFromStart);
file.Read(buf, samples_can_do * bytes_per_sample * channels);
// Update data
buf = (char*)buf + samples_can_do * bytes_per_sample * channels;
start += samples_can_do;
count -= samples_can_do;
}
index++;
}
// If we exhausted all sample sections zerofill the rest
if (count > 0) {
if (bytes_per_sample == 1)
// 8 bit formats are usually unsigned with bias 127
memset(buf, 127, count*channels);
else
// While everything else is signed
memset(buf, 0, count*bytes_per_sample*channels);
}
}
// RIFF WAV PCM provider
// Overview of RIFF WAV: <http://www.sonicspot.com/guide/wavefiles.html>
class RiffWavPCMAudioProvider : public PCMAudioProvider {
private:
struct ChunkHeader {
char type[4];
uint32_t size; // XXX: Assume we're compiling on little endian
};
struct RIFFChunk {
ChunkHeader ch;
char format[4];
};
struct fmtChunk {
// Skip the chunk header here, it's processed separately
uint16_t compression; // compression format used -- 0x01 = PCM
uint16_t channels;
uint32_t samplerate;
uint32_t avg_bytes_sec; // can't always be trusted
uint16_t block_align;
uint16_t significant_bits_sample;
// Here was supposed to be some more fields but we don't need them
// and just skipping by the size of the struct wouldn't be safe
// either way, as the fields can depend on the compression.
};
public:
RiffWavPCMAudioProvider(const wxString &_filename)
{
filename = _filename;
if (!file.Open(_filename, wxFile::read)) throw _T("RIFF PCM WAV audio provider: Unable to open file for reading");
// Read header
file.Seek(0);
RIFFChunk header;
if (file.Read(&header, sizeof(header)) < sizeof(header)) throw _T("RIFF PCM WAV audio provider: file is too small to contain a RIFF header");
// Check that it's good
if (strncmp(header.ch.type, "RIFF", 4)) throw _T("RIFF PCM WAV audio provider: File is not a RIFF file");
if (strncmp(header.format, "WAVE", 4)) throw _T("RIFF PCM WAV audio provider: File is not a RIFF WAV file");
// Count how much more data we can have in the entire file
// The first 4 bytes are already eaten by the header.format field
uint32_t data_left = header.ch.size - 4;
// How far into the file we have processed.
// Must be incremented by the riff chunk size fields.
uint32_t filepos = sizeof(header);
bool got_fmt_header = false;
// Inherited from AudioProvider
num_samples = 0;
// Continue reading chunks until out of data
while (data_left) {
file.Seek(filepos);
ChunkHeader ch;
if (file.Read(&ch, sizeof(ch)) < sizeof(ch)) break;
// Update counters
data_left -= sizeof(ch);
filepos += sizeof(ch);
if (strncmp(ch.type, "fmt ", 4) == 0) {
if (got_fmt_header) throw _T("RIFF PCM WAV audio provider: Invalid file, multiple 'fmt ' chunks");
got_fmt_header = true;
fmtChunk fmt;
if (file.Read(&fmt, sizeof(fmt)) < sizeof(fmt)) throw _T("RIFF PCM WAV audio provider: File ended before end of 'fmt ' chunk");
if (fmt.compression != 1) throw _T("RIFF PCM WAV audio provider: Can't use file, not PCM encoding");
// Set stuff inherited from the AudioProvider class
sample_rate = fmt.samplerate;
channels = fmt.channels;
bytes_per_sample = (fmt.significant_bits_sample + 7) / 8; // round up to nearest whole byte
}
else if (strncmp(ch.type, "data", 4) == 0) {
// This won't pick up 'data' chunks inside 'wavl' chunks
// since the 'wavl' chunks wrap those.
int64_t samples = ch.size / bytes_per_sample;
int64_t frames = samples / channels;
IndexPoint ip;
ip.start_sample = num_samples;
ip.num_samples = frames;
ip.start_byte = filepos;
index_points.push_back(ip);
num_samples += frames;
}
// Support wavl (wave list) chunks too?
// Update counters
// Make sure they're word aligned
data_left -= (ch.size + 1) & ~1;
filepos += (ch.size + 1) & ~1;
}
}
};
// Mix down any number of channels to mono
class DownmixingAudioProvider : public AudioProvider {
private:
AudioProvider *provider;
int src_channels;
public:
DownmixingAudioProvider(AudioProvider *source)
{
filename = source->GetFilename();
channels = 1; // target
src_channels = source->GetChannels();
num_samples = source->GetNumSamples();
bytes_per_sample = source->GetBytesPerSample();
sample_rate = source->GetSampleRate();
// We now own this
provider = source;
if (!(bytes_per_sample == 1 || bytes_per_sample == 2)) throw _T("Downmixing Audio Provider: Can only downmix 8 and 16 bit audio");
}
~DownmixingAudioProvider()
{
delete provider;
}
void GetAudio(void *buf, int64_t start, int64_t count)
{
if (count == 0) return;
// We can do this ourselves
if (start >= num_samples) {
if (bytes_per_sample == 1)
// 8 bit formats are usually unsigned with bias 127
memset(buf, 127, count);
else
// While everything else is signed
memset(buf, 0, count*bytes_per_sample);
return;
}
// So alloc some temporary memory for this
// Depending on use, this might be made faster by using
// a pre-allocced block of memory...?
char *tmp = new char[count*bytes_per_sample*src_channels];
provider->GetAudio(tmp, start, count);
// Now downmix
// Just average the samples over the channels (really bad if they're out of phase!)
if (bytes_per_sample == 1) {
uint8_t *src = (uint8_t *)tmp;
uint8_t *dst = (uint8_t *)buf;
while (count > 0) {
int sum = 0;
for (int c = 0; c < src_channels; c++)
sum += *(src++);
*(dst++) = (uint8_t)(sum / src_channels);
count--;
}
}
else if (bytes_per_sample == 2) {
int16_t *src = (int16_t *)tmp;
int16_t *dst = (int16_t *)buf;
while (count > 0) {
int sum = 0;
for (int c = 0; c < src_channels; c++)
sum += *(src++);
*(dst++) = (int16_t)(sum / src_channels);
count--;
}
}
// Done downmixing, free the work buffer
delete[] tmp;
}
};
AudioProvider *CreatePCMAudioProvider(const wxString &filename)
{
AudioProvider *provider = 0;
// Try Microsoft/IBM RIFF WAV first
// XXX: This is going to blow up if built on big endian archs
try { provider = new RiffWavPCMAudioProvider(filename); }
catch (...) { provider = 0; }
if (provider && provider->GetChannels() > 1) {
// Can't feed non-mono audio to the rest of the program.
// Create a downmixing proxy and if it fails, don't provide PCM.
try {
provider = new DownmixingAudioProvider(provider);
}
catch (...) {
delete provider;
provider = 0;
}
}
return provider;
}