/* * Interface to libmp3lame for mp3 encoding * Copyright (c) 2002 Lennert Buytenhek * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg 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 * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file mp3lameaudio.c * Interface to libmp3lame for mp3 encoding. */ #include "avcodec.h" #include "mpegaudio.h" #include #define BUFFER_SIZE (7200 + MPA_FRAME_SIZE + MPA_FRAME_SIZE/4) typedef struct Mp3AudioContext { lame_global_flags *gfp; int stereo; uint8_t buffer[BUFFER_SIZE]; int buffer_index; } Mp3AudioContext; static av_cold int MP3lame_encode_init(AVCodecContext *avctx) { Mp3AudioContext *s = avctx->priv_data; if (avctx->channels > 2) return -1; s->stereo = avctx->channels > 1 ? 1 : 0; if ((s->gfp = lame_init()) == NULL) goto err; lame_set_in_samplerate(s->gfp, avctx->sample_rate); lame_set_out_samplerate(s->gfp, avctx->sample_rate); lame_set_num_channels(s->gfp, avctx->channels); if(avctx->compression_level == FF_COMPRESSION_DEFAULT) { lame_set_quality(s->gfp, 5); } else { lame_set_quality(s->gfp, avctx->compression_level); } /* lame 3.91 doesn't work in mono */ lame_set_mode(s->gfp, JOINT_STEREO); lame_set_brate(s->gfp, avctx->bit_rate/1000); if(avctx->flags & CODEC_FLAG_QSCALE) { lame_set_brate(s->gfp, 0); lame_set_VBR(s->gfp, vbr_default); lame_set_VBR_q(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA); } lame_set_bWriteVbrTag(s->gfp,0); lame_set_disable_reservoir(s->gfp, avctx->flags2 & CODEC_FLAG2_BIT_RESERVOIR ? 0 : 1); if (lame_init_params(s->gfp) < 0) goto err_close; avctx->frame_size = lame_get_framesize(s->gfp); avctx->coded_frame= avcodec_alloc_frame(); avctx->coded_frame->key_frame= 1; return 0; err_close: lame_close(s->gfp); err: return -1; } static const int sSampleRates[3] = { 44100, 48000, 32000 }; static const int sBitRates[2][3][15] = { { { 0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448}, { 0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384}, { 0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320} }, { { 0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256}, { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}, { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160} }, }; static const int sSamplesPerFrame[2][3] = { { 384, 1152, 1152 }, { 384, 1152, 576 } }; static const int sBitsPerSlot[3] = { 32, 8, 8 }; static int mp3len(void *data, int *samplesPerFrame, int *sampleRate) { uint32_t header = AV_RB32(data); int layerID = 3 - ((header >> 17) & 0x03); int bitRateID = ((header >> 12) & 0x0f); int sampleRateID = ((header >> 10) & 0x03); int bitsPerSlot = sBitsPerSlot[layerID]; int isPadded = ((header >> 9) & 0x01); static int const mode_tab[4]= {2,3,1,0}; int mode= mode_tab[(header >> 19) & 0x03]; int mpeg_id= mode>0; int temp0, temp1, bitRate; if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) { return -1; } if(!samplesPerFrame) samplesPerFrame= &temp0; if(!sampleRate ) sampleRate = &temp1; // *isMono = ((header >> 6) & 0x03) == 0x03; *sampleRate = sSampleRates[sampleRateID]>>mode; bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000; *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID]; //av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode); return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded; } static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame, int buf_size, void *data) { Mp3AudioContext *s = avctx->priv_data; int len; int lame_result; /* lame 3.91 dies on '1-channel interleaved' data */ if(data){ if (s->stereo) { lame_result = lame_encode_buffer_interleaved( s->gfp, data, avctx->frame_size, s->buffer + s->buffer_index, BUFFER_SIZE - s->buffer_index ); } else { lame_result = lame_encode_buffer( s->gfp, data, data, avctx->frame_size, s->buffer + s->buffer_index, BUFFER_SIZE - s->buffer_index ); } }else{ lame_result= lame_encode_flush( s->gfp, s->buffer + s->buffer_index, BUFFER_SIZE - s->buffer_index ); } if(lame_result < 0){ if(lame_result==-1) { /* output buffer too small */ av_log(avctx, AV_LOG_ERROR, "lame: output buffer too small (buffer index: %d, free bytes: %d)\n", s->buffer_index, BUFFER_SIZE - s->buffer_index); } return -1; } s->buffer_index += lame_result; if(s->buffer_index<4) return 0; len= mp3len(s->buffer, NULL, NULL); //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index); if(len <= s->buffer_index){ memcpy(frame, s->buffer, len); s->buffer_index -= len; memmove(s->buffer, s->buffer+len, s->buffer_index); //FIXME fix the audio codec API, so we do not need the memcpy() /*for(i=0; ipriv_data; av_freep(&avctx->coded_frame); lame_close(s->gfp); return 0; } AVCodec libmp3lame_encoder = { "libmp3lame", CODEC_TYPE_AUDIO, CODEC_ID_MP3, sizeof(Mp3AudioContext), MP3lame_encode_init, MP3lame_encode_frame, MP3lame_encode_close, .capabilities= CODEC_CAP_DELAY, .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, .long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"), };