AudioFileMp3: provide support for mp3 export
Add MP3 as one of the options to export the project as. Currently the lib path is hard-coded, need to add that to settings. It loads lame dynamically.
This commit is contained in:
Andrew Kelley
312
src/core/audio/audio_file_mp3.cpp
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312
src/core/audio/audio_file_mp3.cpp
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@@ -0,0 +1,312 @@
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/*
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* audo_file_mp3.cpp - audio-device which encodes mp3-stream and writes it
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* into an mp3-file. This is used for song-export.
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*
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* Copyright (c) 2004-2009 Tobias Doerffel <tobydox/at/users.sourceforge.net>
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* 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 "lame.h"
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#include "audio_file_mp3.h"
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#include <cstring>
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#include <limits>
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using namespace std;
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AudioFileMp3::AudioFileMp3( const sample_rate_t _sample_rate,
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const ch_cnt_t _channels, bool & _success_ful,
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const QString & _file,
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const bool _use_vbr,
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const bitrate_t _nom_bitrate,
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const bitrate_t _min_bitrate,
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const bitrate_t _max_bitrate,
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const int _depth,
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mixer * _mixer ) :
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audioFileDevice( _sample_rate, _channels, _file, _use_vbr,
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_nom_bitrate, _min_bitrate, _max_bitrate,
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_depth, _mixer ),
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m_lgf( NULL ),
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m_lame( NULL ),
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m_outfile( NULL ),
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m_hq_mode( false )
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{
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// connect to lame
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m_ok = initLame("/usr/lib/libmp3lame.so.0");
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m_ok = _success_ful = m_ok && startEncoding();
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}
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bool AudioFileMp3::initLame(QString libpath)
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{
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// dynamically load the .so file
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m_lame = new QLibrary(libpath);
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if( ! m_lame->load() ) return false;
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// grab the functions and stuff we need
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lame_init = (lame_init_t *)
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m_lame->resolve("lame_init");
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get_lame_version = (get_lame_version_t *)
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m_lame->resolve("get_lame_version");
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lame_init_params = (lame_init_params_t *)
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m_lame->resolve("lame_init_params");
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lame_encode_buffer = (lame_encode_buffer_t *)
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m_lame->resolve("lame_encode_buffer");
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lame_encode_buffer_interleaved = (lame_encode_buffer_interleaved_t *)
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m_lame->resolve("lame_encode_buffer_interleaved");
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lame_encode_flush = (lame_encode_flush_t *)
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m_lame->resolve("lame_encode_flush");
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lame_close = (lame_close_t *)
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m_lame->resolve("lame_close");
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lame_set_in_samplerate = (lame_set_in_samplerate_t *)
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m_lame->resolve("lame_set_in_samplerate");
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lame_set_out_samplerate = (lame_set_out_samplerate_t *)
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m_lame->resolve("lame_set_out_samplerate");
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lame_set_num_channels = (lame_set_num_channels_t *)
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m_lame->resolve("lame_set_num_channels");
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lame_set_quality = (lame_set_quality_t *)
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m_lame->resolve("lame_set_quality");
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lame_set_brate = (lame_set_brate_t *)
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m_lame->resolve("lame_set_brate");
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lame_set_VBR = (lame_set_VBR_t *)
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m_lame->resolve("lame_set_VBR");
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lame_set_VBR_q = (lame_set_VBR_q_t *)
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m_lame->resolve("lame_set_VBR_q");
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lame_set_VBR_min_bitrate_kbps = (lame_set_VBR_min_bitrate_kbps_t *)
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m_lame->resolve("lame_set_VBR_min_bitrate_kbps");
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lame_set_mode = (lame_set_mode_t *)
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m_lame->resolve("lame_set_mode");
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lame_set_preset = (lame_set_preset_t *)
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m_lame->resolve("lame_set_preset");
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lame_set_error_protection = (lame_set_error_protection_t *)
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m_lame->resolve("lame_set_error_protection");
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lame_set_disable_reservoir = (lame_set_disable_reservoir_t *)
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m_lame->resolve("lame_set_disable_reservoir");
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lame_set_padding_type = (lame_set_padding_type_t *)
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m_lame->resolve("lame_set_padding_type");
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lame_set_bWriteVbrTag = (lame_set_bWriteVbrTag_t *)
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m_lame->resolve("lame_set_bWriteVbrTag");
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lame_set_findReplayGain = (lame_set_findReplayGain_t *)
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m_lame->resolve("lame_set_findReplayGain");
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lame_set_VBR_quality = (lame_set_VBR_quality_t *)
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m_lame->resolve("lame_set_VBR_quality");
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lame_set_VBR_mean_bitrate_kbps = (lame_set_VBR_mean_bitrate_kbps_t *)
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m_lame->resolve("lame_set_VBR_mean_bitrate_kbps");
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lame_set_VBR_max_bitrate_kbps = (lame_set_VBR_max_bitrate_kbps_t *)
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m_lame->resolve("lame_set_VBR_max_bitrate_kbps");
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// These are optional
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lame_get_lametag_frame = (lame_get_lametag_frame_t *)
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m_lame->resolve("lame_get_lametag_frame");
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lame_mp3_tags_fid = (lame_mp3_tags_fid_t *)
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m_lame->resolve("lame_mp3_tags_fid");
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if (!lame_init ||
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!get_lame_version ||
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!lame_init_params ||
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!lame_encode_buffer ||
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!lame_encode_buffer_interleaved ||
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!lame_encode_flush ||
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!lame_close ||
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!lame_set_in_samplerate ||
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!lame_set_out_samplerate ||
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!lame_set_num_channels ||
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!lame_set_quality ||
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!lame_set_brate ||
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!lame_set_VBR ||
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!lame_set_VBR_q ||
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!lame_set_mode ||
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!lame_set_preset ||
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!lame_set_error_protection ||
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!lame_set_disable_reservoir ||
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!lame_set_padding_type ||
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!lame_set_bWriteVbrTag)
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{
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// some symbols are missing
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printf("AudioFileMp3: some symbols are missing from the lame library\n");
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m_lame->unload();
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return false;
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}
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return true;
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}
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AudioFileMp3::~AudioFileMp3()
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{
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finishEncoding();
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if( m_lame )
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{
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if( m_lame->isLoaded() )
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m_lame->unload();
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delete m_lame;
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m_lame = NULL;
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}
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if( m_outfile )
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{
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if( m_outfile->isOpen() )
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m_outfile->close();
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delete m_outfile;
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}
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}
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void AudioFileMp3::finishEncoding( void )
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{
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if( m_lgf )
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{
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// flush
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int bufSize = 7200;
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unsigned char * out = new unsigned char[bufSize];
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int rc = lame_encode_flush(m_lgf, out, bufSize);
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if( m_outfile && m_outfile->isOpen() ){
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m_outfile->write( (const char *) out, rc );
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m_outfile->close();
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delete m_outfile;
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m_outfile = NULL;
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}
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// cleanup
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delete[] out;
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// close any open handles we may have
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lame_close(m_lgf);
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m_lgf = NULL;
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}
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}
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bool AudioFileMp3::startEncoding( void )
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{
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// open any handles, files, etc
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m_lgf = lame_init();
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if( m_lgf == NULL ){
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printf("AudioFileMp3: Unable to initialize lame\n");
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return false;
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}
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if( channels() > 2 )
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printf("I don't think lame can do more than 2 channels\n");
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lame_set_in_samplerate(m_lgf, sampleRate() );
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lame_set_num_channels(m_lgf, channels() );
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if( m_hq_mode )
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lame_set_quality(m_lgf, 0); // best, very slow
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else
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lame_set_quality(m_lgf, 2); // near-best, not too slow
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lame_set_mode(m_lgf, STEREO);
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lame_set_findReplayGain(m_lgf, 1); // perform ReplayGain analysis
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if( useVBR() == 0 )
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{
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lame_set_VBR(m_lgf, vbr_off);
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lame_set_brate(m_lgf, nominalBitrate() );
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}
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else
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{
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lame_set_VBR(m_lgf, vbr_abr);
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lame_set_VBR_quality(m_lgf, 2);
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lame_set_VBR_mean_bitrate_kbps(m_lgf, nominalBitrate() );
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lame_set_VBR_min_bitrate_kbps(m_lgf, minBitrate() );
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lame_set_VBR_max_bitrate_kbps(m_lgf, maxBitrate() );
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}
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if( lame_init_params( m_lgf ) < 0 )
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return false;
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// open the file
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m_outfile = new QFile( outputFile() );
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if( ! m_outfile->open( QIODevice::WriteOnly ) )
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{
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printf("AudioFileMp3: unable to open file for output\n");
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return false;
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}
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// write the headers and such
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return true;
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}
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short int AudioFileMp3::rescale(float sample) {
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return (sample / 1) * std::numeric_limits<short int>::max();
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}
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// encode data and write to file
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void AudioFileMp3::writeBuffer( const surroundSampleFrame * _ab,
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const fpp_t _frames, const float _master_gain )
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{
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// encode with lame
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int bufSize = 1.25 * _frames + 7200;
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short int * in = new short int[_frames*2];
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unsigned char * out = new unsigned char[bufSize];
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// scale to short int instead of float
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for(int i=0; i < _frames; ++i)
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{
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in[i*2] = rescale( _ab[i][0] );
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in[i*2+1] = rescale( _ab[i][1] );
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}
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int rc = lame_encode_buffer_interleaved( m_lgf, in, _frames, out, bufSize);
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switch(rc){
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case -1:
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printf("AudioFileMp3: encode error: buffer too small.\n");
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return;
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case -2:
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printf("AudioFileMp3: encode error: out of memory\n");
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return;
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case -3:
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printf("AudioFileMp3: encode error: lame_init_params not called\n");
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return;
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case -4:
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printf("AudioFileMp3: encode error: psycho acoustic problems\n");
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return;
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}
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// write to file
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m_outfile->write( (const char *) out, rc );
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// clean up
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delete[] out;
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delete[] in;
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}
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@@ -162,7 +162,7 @@ int main( int argc, char * * argv )
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"-r, --render <project file> render given project file\n"
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"-o, --output <file> render into <file>\n"
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"-f, --output-format <format> specify format of render-output where\n"
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" format is either 'wav' or 'ogg'.\n"
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" format is either 'wav', 'ogg', or 'mp3'.\n"
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"-s, --samplerate <samplerate> specify output samplerate in Hz\n"
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" range: 44100 (default) to 192000\n"
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"-b, --bitrate <bitrate> specify output bitrate in kHz\n"
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@@ -227,6 +227,10 @@ int main( int argc, char * * argv )
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eff = projectRenderer::OggFile;
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}
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#endif
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else if( ext == "mp3" )
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{
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eff = projectRenderer::Mp3File;
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}
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else
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{
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printf( "\nInvalid output format %s.\n\n"
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@@ -491,10 +495,7 @@ int main( int argc, char * * argv )
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{
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// create renderer
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projectRenderer * r = new projectRenderer( qs, os, eff,
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render_out +
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QString( ( eff ==
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projectRenderer::WaveFile ) ?
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"wav" : "ogg" ) );
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render_out + QString(projectRenderer::EFF_ext[eff]));
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QCoreApplication::instance()->connect( r,
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SIGNAL( finished() ), SLOT( quit() ) );
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@@ -31,6 +31,7 @@
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#include "audio_file_wave.h"
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#include "audio_file_ogg.h"
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#include "audio_file_mp3.h"
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#ifdef LMMS_HAVE_PTHREAD_H
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#include <pthread.h>
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@@ -53,6 +54,9 @@ fileEncodeDevice __fileEncodeDevices[] =
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NULL
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#endif
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},
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{ projectRenderer::Mp3File,
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QT_TRANSLATE_NOOP( "projectRenderer", "MP3 File (*.mp3)" ),
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".mp3", &AudioFileMp3::getInst },
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// ... insert your own file-encoder-infos here... may be one day the
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// user can add own encoders inside the program...
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@@ -60,7 +64,7 @@ fileEncodeDevice __fileEncodeDevices[] =
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} ;
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const char * projectRenderer::EFF_ext[] = {"wav", "ogg", "mp3"};
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projectRenderer::projectRenderer( const mixer::qualitySettings & _qs,
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Reference in New Issue
Block a user