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Diffstat (limited to 'src/liblame_wrapper.cc')
| -rw-r--r-- | src/liblame_wrapper.cc | 293 |
1 files changed, 0 insertions, 293 deletions
diff --git a/src/liblame_wrapper.cc b/src/liblame_wrapper.cc deleted file mode 100644 index 5603d6f..0000000 --- a/src/liblame_wrapper.cc +++ /dev/null @@ -1,293 +0,0 @@ -/* -*- 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, 1152); - // lame_set_num_samples(gfp, SAMPLES); - // lame_set_num_samples(gfp, 0); - - 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; - - calc_bitrate = 0; - frame_number = 0; -} - -LibLAMEWrapper::~LibLAMEWrapper() -{ - delete audio_buffer[0]; - delete audio_buffer[1]; -} - -Frame *LibLAMEWrapper::close(Frame *oldframe) -{ - Frame *frame; - unsigned int offset = 0; - - frame = new Frame(NULL, (int)(1.25 * SAMPLES + 7200) * 2); // Big enough to hold two frames - - if(oldframe) { - offset = oldframe->size; - frame->number = oldframe->number; - memcpy(frame->data, oldframe->data, oldframe->size); - delete oldframe; - } - - int flush; - - flush = lame_encode_finish(gfp, frame->data + offset, 7200); - - frame->size = offset + flush; - - calc_bitrate += flush; - frame->bitrate = (unsigned int)((double)calc_bitrate / (double)(frame_number)) * 25; - - return frame; -} - -#include <math.h> -static unsigned int sin_cnt = 0; -Frame *LibLAMEWrapper::encode(Frame *dvframe) -{ - if(dvframe->mute) { - // Overwrite audiobuffer with dummy data - double volume = 1000; // Min:= 0 - Max := 32000 - double frequency = 440; // in Hz - - for(int cnt = 0; cnt < SAMPLES; cnt++) { - sin_cnt++; - double sin_val = (((double)sin_cnt / (double)OUTPUT_SAMPLE_RATE) * (double)M_PI) * frequency; - audio_buffer[0][cnt] = audio_buffer[1][cnt] = (short int)(sin(sin_val) * volume); - } - - // memset(audio_buffer[0], 0, sizeof(audio_buffer[0])); - // memset(audio_buffer[1], 0, sizeof(audio_buffer[1])); - } else { - // Decode audio from dv frame - 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 = 0; - - /* - 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 - */ - - // info->info("VAL: %d - FLUSH_SZ: %d - TOTAL: %d", val, flush_sz, (val + flush_sz)); - - audio_frame->size = val + flush_sz; - - /* - - int bitrate_kbps[14]; - // lame_bitrate_kbps(gfp, bitrate_kbps); - lame_bitrate_hist(gfp, bitrate_kbps); - // 32 40 48 56 64 80 96 112 128 160 192 224 256 320 - info->info("%d %d %d %d %d %d %d %d %d %d %d %d %d %d", - bitrate_kbps[0], - bitrate_kbps[1], - bitrate_kbps[2], - bitrate_kbps[3], - bitrate_kbps[4], - bitrate_kbps[5], - bitrate_kbps[6], - bitrate_kbps[7], - bitrate_kbps[8], - bitrate_kbps[9], - bitrate_kbps[10], - bitrate_kbps[11], - bitrate_kbps[12], - bitrate_kbps[13]); - */ - // while(frame_number != lame_get_frameNum(gfp)) { - - calc_bitrate += audio_frame->size;//lame_get_framesize(gfp); - frame_number ++;//= 1;//lame_get_frameNum(gfp); - - // info->info("lame_get_frameNum(gfp) %d ?= frame_number %d", lame_get_frameNum(gfp), frame_number); - // } - - // Bits pr. second - // 25 * 7 frames pr.second (it seems!) - audio_frame->bitrate = (unsigned int)((double)calc_bitrate / (double)(frame_number)) * 25; - /* - info->info("Audio size: %d, bitrate: %.4f", - audio_frame->bitrate, - (float)(config->readInt("mp3_bitrate") * 1000)/(float)(audio_frame->bitrate)); - */ - - /* - FILE* fp = fopen("/tmp/audiotest.mp3", "a"); - fwrite(audio_frame->data, audio_frame->size, 1, fp); - fclose(fp); - */ - return audio_frame; -} |