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/*
Rosegarden
A sequencer and musical notation editor.
This program is Copyright 2000-2008
Guillaume Laurent <glaurent@telegraph-road.org>,
Chris Cannam <cannam@all-day-breakfast.com>,
Richard Bown <bownie@bownie.com>
The moral right of the authors to claim authorship of this work
has been asserted.
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License as
published by the Free Software Foundation; either version 2 of the
License, or (at your option) any later version. See the file
COPYING included with this distribution for more information.
*/
#include "RecordableAudioFile.h"
#include <cstdlib>
//#define DEBUG_RECORDABLE 1
namespace Rosegarden
{
RecordableAudioFile::RecordableAudioFile(AudioFile *audioFile,
size_t bufferSize) :
m_audioFile(audioFile),
m_status(IDLE)
{
for (unsigned int ch = 0; ch < audioFile->getChannels(); ++ch) {
m_ringBuffers.push_back(new RingBuffer<sample_t>(bufferSize));
if (!m_ringBuffers[ch]->mlock()) {
std::cerr << "WARNING: RecordableAudioFile::initialise: couldn't lock buffer into real memory, performance may be impaired" << std::endl;
}
}
}
RecordableAudioFile::~RecordableAudioFile()
{
write();
m_audioFile->close();
delete m_audioFile;
for (size_t i = 0; i < m_ringBuffers.size(); ++i) {
delete m_ringBuffers[i];
}
}
size_t
RecordableAudioFile::buffer(const sample_t *data, int channel, size_t frames)
{
if (channel >= int(m_ringBuffers.size())) {
std::cerr << "RecordableAudioFile::buffer: No such channel as "
<< channel << std::endl;
return 0;
}
size_t available = m_ringBuffers[channel]->getWriteSpace();
if (frames > available) {
std::cerr << "RecordableAudioFile::buffer: buffer maxed out!" << std::endl;
frames = available;
}
#ifdef DEBUG_RECORDABLE
std::cerr << "RecordableAudioFile::buffer: buffering " << frames << " frames on channel " << channel << std::endl;
#endif
m_ringBuffers[channel]->write(data, frames);
return frames;
}
void
RecordableAudioFile::write()
{
// Use a static buffer -- this obviously requires that write() is
// only called from a single thread
static size_t bufferSize = 0;
static sample_t *buffer = 0;
static char *encodeBuffer = 0;
unsigned int bits = m_audioFile->getBitsPerSample();
if (bits != 16 && bits != 32) {
std::cerr << "ERROR: RecordableAudioFile::write: file has " << bits
<< " bits per sample; only 16 or 32 are supported" << std::endl;
return ;
}
unsigned int channels = m_audioFile->getChannels();
unsigned char b1, b2;
// We need the same amount of available data on every channel
size_t s = 0;
for (unsigned int ch = 0; ch < channels; ++ch) {
size_t available = m_ringBuffers[ch]->getReadSpace();
#ifdef DEBUG_RECORDABLE
std::cerr << "RecordableAudioFile::write: " << available << " frames available to write on channel " << ch << std::endl;
#endif
if (ch == 0 || available < s)
s = available;
}
if (s == 0)
return ;
size_t bufferReqd = channels * s;
if (bufferReqd > bufferSize) {
if (buffer) {
buffer = (sample_t *)realloc(buffer, bufferReqd * sizeof(sample_t));
encodeBuffer = (char *)realloc(encodeBuffer, bufferReqd * 4);
} else {
buffer = (sample_t *) malloc(bufferReqd * sizeof(sample_t));
encodeBuffer = (char *)malloc(bufferReqd * 4);
}
bufferSize = bufferReqd;
}
for (unsigned int ch = 0; ch < channels; ++ch) {
m_ringBuffers[ch]->read(buffer + ch * s, s);
}
// interleave and convert
if (bits == 16) {
size_t index = 0;
for (size_t i = 0; i < s; ++i) {
for (unsigned int ch = 0; ch < channels; ++ch) {
float sample = buffer[i + ch * s];
b2 = (unsigned char)((long)(sample * 32767.0) & 0xff);
b1 = (unsigned char)((long)(sample * 32767.0) >> 8);
encodeBuffer[index++] = b2;
encodeBuffer[index++] = b1;
}
}
} else {
char *encodePointer = encodeBuffer;
for (size_t i = 0; i < s; ++i) {
for (unsigned int ch = 0; ch < channels; ++ch) {
float sample = buffer[i + ch * s];
*(float *)encodePointer = sample;
encodePointer += sizeof(float);
}
}
}
#ifdef DEBUG_RECORDABLE
std::cerr << "RecordableAudioFile::write: writing " << s << " frames at " << channels << " channels and " << bits << " bits to file" << std::endl;
#endif
m_audioFile->appendSamples(encodeBuffer, s);
}
}
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