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214 lines
6.2 KiB
214 lines
6.2 KiB
/* |
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* Interface to libmp3lame for mp3 encoding |
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* Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org> |
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* |
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* This library is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser 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 library 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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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with this library; if not, write to the Free Software |
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA |
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*/ |
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/** |
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* @file mp3lameaudio.c |
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* Interface to libmp3lame for mp3 encoding. |
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*/ |
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#include "avcodec.h" |
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#include "mpegaudio.h" |
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#include <lame/lame.h> |
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#define BUFFER_SIZE (2*MPA_FRAME_SIZE) |
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typedef struct Mp3AudioContext { |
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lame_global_flags *gfp; |
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int stereo; |
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uint8_t buffer[BUFFER_SIZE]; |
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int buffer_index; |
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} Mp3AudioContext; |
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static int MP3lame_encode_init(AVCodecContext *avctx) |
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{ |
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Mp3AudioContext *s = avctx->priv_data; |
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if (avctx->channels > 2) |
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return -1; |
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s->stereo = avctx->channels > 1 ? 1 : 0; |
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if ((s->gfp = lame_init()) == NULL) |
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goto err; |
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lame_set_in_samplerate(s->gfp, avctx->sample_rate); |
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lame_set_out_samplerate(s->gfp, avctx->sample_rate); |
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lame_set_num_channels(s->gfp, avctx->channels); |
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/* lame 3.91 dies on quality != 5 */ |
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lame_set_quality(s->gfp, 5); |
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/* lame 3.91 doesn't work in mono */ |
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lame_set_mode(s->gfp, JOINT_STEREO); |
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lame_set_brate(s->gfp, avctx->bit_rate/1000); |
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lame_set_bWriteVbrTag(s->gfp,0); |
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if (lame_init_params(s->gfp) < 0) |
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goto err_close; |
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avctx->frame_size = lame_get_framesize(s->gfp); |
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avctx->coded_frame= avcodec_alloc_frame(); |
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avctx->coded_frame->key_frame= 1; |
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return 0; |
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err_close: |
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lame_close(s->gfp); |
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err: |
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return -1; |
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} |
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static const int sSampleRates[3] = { |
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44100, 48000, 32000 |
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}; |
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static const int sBitRates[2][3][15] = { |
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{ { 0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448}, |
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{ 0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384}, |
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{ 0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320} |
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}, |
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{ { 0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256}, |
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{ 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}, |
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{ 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160} |
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}, |
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}; |
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static const int sSamplesPerFrame[2][3] = |
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{ |
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{ 384, 1152, 1152 }, |
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{ 384, 1152, 576 } |
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}; |
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static const int sBitsPerSlot[3] = { |
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32, |
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8, |
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8 |
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}; |
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static int mp3len(void *data, int *samplesPerFrame, int *sampleRate) |
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{ |
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uint8_t *dataTmp = (uint8_t *)data; |
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uint32_t header = ( (uint32_t)dataTmp[0] << 24 ) | ( (uint32_t)dataTmp[1] << 16 ) | ( (uint32_t)dataTmp[2] << 8 ) | (uint32_t)dataTmp[3]; |
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int layerID = 3 - ((header >> 17) & 0x03); |
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int bitRateID = ((header >> 12) & 0x0f); |
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int sampleRateID = ((header >> 10) & 0x03); |
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int bitsPerSlot = sBitsPerSlot[layerID]; |
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int isPadded = ((header >> 9) & 0x01); |
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static int const mode_tab[4]= {2,3,1,0}; |
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int mode= mode_tab[(header >> 19) & 0x03]; |
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int mpeg_id= mode>0; |
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int temp0, temp1, bitRate; |
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if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) { |
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return -1; |
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} |
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if(!samplesPerFrame) samplesPerFrame= &temp0; |
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if(!sampleRate ) sampleRate = &temp1; |
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// *isMono = ((header >> 6) & 0x03) == 0x03; |
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*sampleRate = sSampleRates[sampleRateID]>>mode; |
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bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000; |
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*samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID]; |
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//av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode); |
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return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded; |
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} |
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int MP3lame_encode_frame(AVCodecContext *avctx, |
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unsigned char *frame, int buf_size, void *data) |
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{ |
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Mp3AudioContext *s = avctx->priv_data; |
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int len, i; |
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int lame_result; |
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/* lame 3.91 dies on '1-channel interleaved' data */ |
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if(data){ |
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if (s->stereo) { |
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lame_result = lame_encode_buffer_interleaved( |
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s->gfp, |
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data, |
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avctx->frame_size, |
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s->buffer + s->buffer_index, |
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BUFFER_SIZE - s->buffer_index |
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); |
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} else { |
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lame_result = lame_encode_buffer( |
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s->gfp, |
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data, |
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data, |
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avctx->frame_size, |
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s->buffer + s->buffer_index, |
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BUFFER_SIZE - s->buffer_index |
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); |
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} |
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}else{ |
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lame_result= lame_encode_flush( |
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s->gfp, |
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s->buffer + s->buffer_index, |
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BUFFER_SIZE - s->buffer_index |
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); |
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} |
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if(lame_result==-1) { |
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/* output buffer too small */ |
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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); |
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return 0; |
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} |
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s->buffer_index += lame_result; |
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if(s->buffer_index<4) |
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return 0; |
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len= mp3len(s->buffer, NULL, NULL); |
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//av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index); |
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if(len <= s->buffer_index){ |
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memcpy(frame, s->buffer, len); |
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s->buffer_index -= len; |
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memmove(s->buffer, s->buffer+len, s->buffer_index); |
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//FIXME fix the audio codec API, so we dont need the memcpy() |
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/*for(i=0; i<len; i++){ |
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av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]); |
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}*/ |
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return len; |
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}else |
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return 0; |
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} |
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int MP3lame_encode_close(AVCodecContext *avctx) |
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{ |
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Mp3AudioContext *s = avctx->priv_data; |
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av_freep(&avctx->coded_frame); |
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lame_close(s->gfp); |
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return 0; |
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} |
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AVCodec mp3lame_encoder = { |
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"mp3", |
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CODEC_TYPE_AUDIO, |
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CODEC_ID_MP3, |
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sizeof(Mp3AudioContext), |
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MP3lame_encode_init, |
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MP3lame_encode_frame, |
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MP3lame_encode_close, |
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.capabilities= CODEC_CAP_DELAY, |
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};
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