Tobias Doerffel
20589f19e4
Until now, Mixer not only was responsible for rendering audio buffers but also managed writing them to audio backend (through a FIFO) and handled various quality related parameters. All this functionality has been moved into the new AudioOutputContext class. It glues together AudioBackend (formerly called AudioDevice), global quality settings and the Mixer. The AudioOutputContext class creates a FifoWriter which calls Mixer::renderNextBuffer() and writes the output into the BufferFifo of the AudioOutputContext it belongs to. The BufferFifo is read by the according AudioBackend which belongs to the AudioOutputContext as well. The AudioOutputContext also handles resampling in case the AudioBackend wants the buffers in a different samplerate. During this rewrite the Mixer class and the according source files have been renamed from "mixer" to "Mixer". This results in small changes all over LMMS' code base.
130 lines
3.3 KiB
C++
130 lines
3.3 KiB
C++
/*
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* AudioFileFlac.cpp - Audio-device which encodes a flac stream and writes it
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* into a flac file. This is used for song-export.
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*
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* Copyright (c) 2009 Andrew Kelley <superjoe30@gmail.com>
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*
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* This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* General Public License for more details.
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*
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* You should have received a copy of the GNU General Public
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* License along with this program (see COPYING); if not, write to the
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* Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
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* Boston, MA 02110-1301 USA.
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*
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*/
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#include "AudioFileFlac.h"
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#ifdef LMMS_HAVE_FLAC
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#include <QtDebug>
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#include "lmms_basics.h"
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AudioFileFlac::AudioFileFlac( const sample_rate_t _sample_rate,
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const ch_cnt_t _channels, bool & _success_ful, const QString & _file,
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const bool _use_vbr, const bitrate_t _nom_bitrate,
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const bitrate_t _min_bitrate, const bitrate_t _max_bitrate,
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const int _depth, AudioOutputContext * context ) :
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AudioFileDevice( _sample_rate, _channels, _file, _use_vbr, _nom_bitrate,
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_min_bitrate, _max_bitrate, _depth, context )
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{
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_success_ful = startEncoding();
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}
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bool AudioFileFlac::startEncoding()
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{
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// check the encoder
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if( ! m_encoder )
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{
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qWarning() << "AudioFileFlac: unable to allocate encoder";
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return false;
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}
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bool ok = true;
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ok &= m_encoder.set_verify(true);
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ok &= m_encoder.set_compression_level(5);
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ok &= m_encoder.set_channels(channels());
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ok &= m_encoder.set_bits_per_sample(16);
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ok &= m_encoder.set_sample_rate(sampleRate());
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//ok &= encoder.set_total_samples_estimate(??);
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if( ! ok ) return false;
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// TODO: set metadata: http://flac.cvs.sourceforge.net/viewvc/flac/flac/examples/cpp/encode/file/main.cpp?view=markup
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// initialize encoder
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FLAC__StreamEncoderInitStatus init_status = m_encoder.init(
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outputFile().toAscii());
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if( init_status != FLAC__STREAM_ENCODER_INIT_STATUS_OK )
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{
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qWarning() << "AudioFileFlac: unable to initialize encoder:" <<
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FLAC__StreamEncoderInitStatusString[init_status];
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return false;
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}
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return true;
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}
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Sint16 AudioFileFlac::rescale(float sample) {
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return (qMax<float>(qMin<float>(sample, 1), -1) / 1)
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* std::numeric_limits<Sint16>::max();
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}
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// encode data and write to file
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void AudioFileFlac::writeBuffer( const surroundSampleFrame * _ab,
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const fpp_t _frames, const float _master_gain )
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{
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// scale to short int instead of float
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FLAC__int32 * in = new FLAC__int32[_frames*channels()];
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for(int i=0; i < _frames; ++i)
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{
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for(int c=0; c < channels(); ++c)
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{
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in[i*channels()+c] = (FLAC__int32) rescale( _ab[i][c]
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* _master_gain );
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}
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}
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// feed to the encoder
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m_encoder.process_interleaved(in, _frames);
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// clean up
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delete[] in;
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}
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void AudioFileFlac::finishEncoding()
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{
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m_encoder.finish();
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}
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AudioFileFlac::~AudioFileFlac()
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{
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finishEncoding();
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}
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#endif
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/* vim: set tw=0 noexpandtab: */
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