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/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/***************************************************************************
* liblame_wrapper.cc
*
* Sat Jul 2 11:11:34 CEST 2005
* Copyright 2005 Bent Bisballe
* deva@aasimon.org
****************************************************************************/
/*
* This file is part of MIaV.
*
* MIaV 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.
*
* MIaV is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with MIaV; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
*/
#include <config.h>
#include "liblame_wrapper.h"
#include "miav_config.h"
LibLAMEWrapper::LibLAMEWrapper(Info *i)
{
info = i;
// Init library.
if( (gfp = lame_init()) == NULL) {
info->error("LAME initialization failed (due to malloc failure!)");
return;
}
lame_set_in_samplerate(gfp, INPUT_SAMPLE_RATE);
lame_set_out_samplerate(gfp, OUTPUT_SAMPLE_RATE);
lame_set_num_channels(gfp, CHANNELS);
lame_set_num_samples(gfp, SAMPLES);
lame_set_quality(gfp, config->readInt("mp3_quality"));
lame_set_mode(gfp, STEREO);
lame_set_brate(gfp, config->readInt("mp3_bitrate"));
lame_set_strict_ISO(gfp, 1);
// 1 = write a Xing VBR header frame.
lame_set_bWriteVbrTag(gfp, 0);
// Types of VBR. default = vbr_off = CBR
// lame_set_VBR(gfp, vbr_rh);
// VBR quality level. 0=highest 9=lowest
// lame_set_VBR_q(gfp, 6);
lame_set_copyright(gfp, 0); // is there a copyright on the encoded data?
lame_set_original(gfp, 1); // is the encoded data on the original media?
lame_set_error_protection(gfp, 0);// add 2 byte CRC protection to each frame?
lame_set_padding_type(gfp, PAD_NO); // PAD_NO, PAD_ALL, PAD_ADJUST, PAD_MAX_INDICATOR
// 0 = do not pad frames
// 1 = always pad frames
// 2 = adjust padding to satisfy bit rate
lame_set_extension(gfp, 0); // private extension bit
if (lame_init_params(gfp) < 0) {
info->error("LAME parameter initialization failed.");
return;
}
audio_buffer[0] = new int16_t[AUDIO_BUFFER_SIZE];
audio_buffer[1] = new int16_t[AUDIO_BUFFER_SIZE];
// And now for the dv decoder!
decoder = NULL;
}
LibLAMEWrapper::~LibLAMEWrapper()
{
lame_close(gfp);
delete audio_buffer[0];
delete audio_buffer[1];
}
Frame *LibLAMEWrapper::encode(Frame *dvframe)
{
if(!decoder) {
decoder = dv_decoder_new(FALSE/*this value is unused*/, FALSE, FALSE);
decoder->quality = DV_QUALITY_BEST;
dv_parse_header(decoder, dvframe->data);
decoder->system = e_dv_system_625_50; // PAL lines, PAL framerate
decoder->sampling = e_dv_sample_422; // 4 bytes y, 2 bytes u, 2 bytes v
decoder->std = e_dv_std_iec_61834;
decoder->num_dif_seqs = 12;
}
/**
* Decode audio using libdv
*/
dv_decode_full_audio( decoder, dvframe->data, audio_buffer );
/**
* input pcm data, output (maybe) mp3 frames.
* This routine handles all buffering, resampling and filtering for you.
*
* The required mp3buf_size can be computed from num_samples,
* samplerate and encoding rate, but here is a worst case estimate:
*
* return code number of bytes output in mp3buffer. can be 0
* if return code = -1: mp3buffer was too small
*
* mp3buf_size in bytes = 1.25*num_samples + 7200
*
* I think a tighter bound could be: (mt, March 2000)
* MPEG1:
* num_samples*(bitrate/8)/samplerate + 4*1152*(bitrate/8)/samplerate + 512
* MPEG2:
* num_samples*(bitrate/8)/samplerate + 4*576*(bitrate/8)/samplerate + 256
*
* but test first if you use that!
*
* set mp3buf_size = 0 and LAME will not check if mp3buf_size is
* large enough.
*
* NOTE:
* if gfp->num_channels=2, but gfp->mode = 3 (mono), the L & R channels
* will be averaged into the L channel before encoding only the L channel
* This will overwrite the data in buffer_l[] and buffer_r[].
*
*/
Frame* audio_frame = new Frame(NULL, (int)(1.25 * SAMPLES + 7200));
const short int *buffer_l = audio_buffer[0]; // PCM data for left channel
const short int *buffer_r = audio_buffer[1]; // PCM data for right channel
const int nsamples = SAMPLES; // number of samples per channel
unsigned char* mp3buf = audio_frame->data; // pointer to encoded MP3 stream
const int mp3buf_size = audio_frame->size; // number of valid octets in this
int val;
val = lame_encode_buffer(gfp, buffer_l, buffer_r, nsamples, mp3buf, mp3buf_size);
// val = lame_encode_mp3_frame(gfp, buffer_l, buffer_r, mp3buf, mp3buf_size);
// info->info("Framenr: %d", lame_get_frameNum(gfp));
if(val < 0) {
switch(val) {
case -1: // mp3buf was too small
info->error("Lame encoding failed, mp3buf was too small.");
break;
case -2: // malloc() problem
info->error("Lame encoding failed, due to malloc() problem.");
break;
case -3: // lame_init_params() not called
info->error("Lame encoding failed, lame_init_params() not called.");
break;
case -4: // psycho acoustic problems
info->error("Lame encoding failed, due to psycho acoustic problems.");
break;
default:
info->error("Lame encoding failed, due to unknown error.");
break;
}
}
/**
* OPTIONAL:
* lame_encode_flush_nogap will flush the internal mp3 buffers and pad
* the last frame with ancillary data so it is a complete mp3 frame.
*
* 'mp3buf' should be at least 7200 bytes long
* to hold all possible emitted data.
*
* After a call to this routine, the outputed mp3 data is complete, but
* you may continue to encode new PCM samples and write future mp3 data
* to a different file. The two mp3 files will play back with no gaps
* if they are concatenated together.
*
* This routine will NOT write id3v1 tags into the bitstream.
*
* return code = number of bytes output to mp3buf. Can be 0
*/
int flush_sz = lame_encode_flush_nogap(gfp, // global context handle
mp3buf + val, // pointer to encoded MP3 stream
mp3buf_size - val); // number of valid octets in this stream
audio_frame->size = val + flush_sz;
// Bits pr. second
audio_frame->bitrate = config->readInt("mp3_bitrate") * 1000;
return audio_frame;
}
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