diff options
author | Diego Biurrun <diego@biurrun.de> | 2007-06-06 04:14:18 +0400 |
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committer | Diego Biurrun <diego@biurrun.de> | 2007-06-06 04:14:18 +0400 |
commit | 6f1af73557824b1872732b7823e881a14f870077 (patch) | |
tree | 3d22e9a9585b538bd4295898f1e2f5a89d36619b /libavcodec/libmp3lame.c | |
parent | 6f74b71ef0f5457b8cc736cb8f828ec1b75a217c (diff) |
Give all wrappers for external libraries names starting with lib.
Originally committed as revision 9226 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/libmp3lame.c')
-rw-r--r-- | libavcodec/libmp3lame.c | 220 |
1 files changed, 220 insertions, 0 deletions
diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c new file mode 100644 index 0000000000..9a5177af30 --- /dev/null +++ b/libavcodec/libmp3lame.c @@ -0,0 +1,220 @@ +/* + * Interface to libmp3lame for mp3 encoding + * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org> + * + * 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 <lame/lame.h> + +#define BUFFER_SIZE (2*MPA_FRAME_SIZE) +typedef struct Mp3AudioContext { + lame_global_flags *gfp; + int stereo; + uint8_t buffer[BUFFER_SIZE]; + int buffer_index; +} Mp3AudioContext; + +static 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); + /* lame 3.91 dies on quality != 5 */ + lame_set_quality(s->gfp, 5); + /* 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); + 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==-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 0; + } + + 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 dont need the memcpy() +/*for(i=0; i<len; i++){ + av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]); +}*/ + return len; + }else + return 0; +} + +static int MP3lame_encode_close(AVCodecContext *avctx) +{ + Mp3AudioContext *s = avctx->priv_data; + + av_freep(&avctx->coded_frame); + + lame_close(s->gfp); + return 0; +} + + +AVCodec mp3lame_encoder = { + "mp3", + CODEC_TYPE_AUDIO, + CODEC_ID_MP3, + sizeof(Mp3AudioContext), + MP3lame_encode_init, + MP3lame_encode_frame, + MP3lame_encode_close, + .capabilities= CODEC_CAP_DELAY, +}; |