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libavcodec/libmp3lame.c

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00001 /*
00002  * Interface to libmp3lame for mp3 encoding
00003  * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
00004  *
00005  * This file is part of Libav.
00006  *
00007  * Libav is free software; you can redistribute it and/or
00008  * modify it under the terms of the GNU Lesser General Public
00009  * License as published by the Free Software Foundation; either
00010  * version 2.1 of the License, or (at your option) any later version.
00011  *
00012  * Libav is distributed in the hope that it will be useful,
00013  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00014  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00015  * Lesser General Public License for more details.
00016  *
00017  * You should have received a copy of the GNU Lesser General Public
00018  * License along with Libav; if not, write to the Free Software
00019  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00020  */
00021 
00027 #include "libavutil/intreadwrite.h"
00028 #include "libavutil/log.h"
00029 #include "libavutil/opt.h"
00030 #include "avcodec.h"
00031 #include "mpegaudio.h"
00032 #include <lame/lame.h>
00033 
00034 #define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
00035 typedef struct Mp3AudioContext {
00036     AVClass *class;
00037     lame_global_flags *gfp;
00038     int stereo;
00039     uint8_t buffer[BUFFER_SIZE];
00040     int buffer_index;
00041     int reservoir;
00042 } Mp3AudioContext;
00043 
00044 static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
00045 {
00046     Mp3AudioContext *s = avctx->priv_data;
00047 
00048     if (avctx->channels > 2)
00049         return -1;
00050 
00051     s->stereo = avctx->channels > 1 ? 1 : 0;
00052 
00053     if ((s->gfp = lame_init()) == NULL)
00054         goto err;
00055     lame_set_in_samplerate(s->gfp, avctx->sample_rate);
00056     lame_set_out_samplerate(s->gfp, avctx->sample_rate);
00057     lame_set_num_channels(s->gfp, avctx->channels);
00058     if (avctx->compression_level == FF_COMPRESSION_DEFAULT) {
00059         lame_set_quality(s->gfp, 5);
00060     } else {
00061         lame_set_quality(s->gfp, avctx->compression_level);
00062     }
00063     lame_set_mode(s->gfp, s->stereo ? JOINT_STEREO : MONO);
00064     lame_set_brate(s->gfp, avctx->bit_rate / 1000);
00065     if (avctx->flags & CODEC_FLAG_QSCALE) {
00066         lame_set_brate(s->gfp, 0);
00067         lame_set_VBR(s->gfp, vbr_default);
00068         lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
00069     }
00070     lame_set_bWriteVbrTag(s->gfp,0);
00071 #if FF_API_LAME_GLOBAL_OPTS
00072     s->reservoir = avctx->flags2 & CODEC_FLAG2_BIT_RESERVOIR;
00073 #endif
00074     lame_set_disable_reservoir(s->gfp, !s->reservoir);
00075     if (lame_init_params(s->gfp) < 0)
00076         goto err_close;
00077 
00078     avctx->frame_size             = lame_get_framesize(s->gfp);
00079     avctx->coded_frame            = avcodec_alloc_frame();
00080     avctx->coded_frame->key_frame = 1;
00081 
00082     return 0;
00083 
00084 err_close:
00085     lame_close(s->gfp);
00086 err:
00087     return -1;
00088 }
00089 
00090 static const int sSampleRates[] = {
00091     44100, 48000,  32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
00092 };
00093 
00094 static const int sBitRates[2][3][15] = {
00095     {
00096         { 0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448 },
00097         { 0, 32, 48, 56, 64,  80,  96,  112, 128, 160, 192, 224, 256, 320, 384 },
00098         { 0, 32, 40, 48, 56,  64,  80,  96,  112, 128, 160, 192, 224, 256, 320 }
00099     },
00100     {
00101         { 0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256 },
00102         { 0,  8, 16, 24, 32, 40, 48,  56,  64,  80,  96, 112, 128, 144, 160 },
00103         { 0,  8, 16, 24, 32, 40, 48,  56,  64,  80,  96, 112, 128, 144, 160 }
00104     },
00105 };
00106 
00107 static const int sSamplesPerFrame[2][3] = {
00108     { 384, 1152, 1152 },
00109     { 384, 1152,  576 }
00110 };
00111 
00112 static const int sBitsPerSlot[3] = { 32, 8, 8 };
00113 
00114 static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
00115 {
00116     uint32_t header  = AV_RB32(data);
00117     int layerID      = 3 - ((header >> 17) & 0x03);
00118     int bitRateID    = ((header >> 12) & 0x0f);
00119     int sampleRateID = ((header >> 10) & 0x03);
00120     int bitsPerSlot  = sBitsPerSlot[layerID];
00121     int isPadded     = ((header >> 9) & 0x01);
00122     static int const mode_tab[4] = { 2, 3, 1, 0 };
00123     int mode    = mode_tab[(header >> 19) & 0x03];
00124     int mpeg_id = mode > 0;
00125     int temp0, temp1, bitRate;
00126 
00127     if (((header >> 21) & 0x7ff) != 0x7ff || mode == 3 || layerID == 3 ||
00128         sampleRateID == 3) {
00129         return -1;
00130     }
00131 
00132     if (!samplesPerFrame)
00133         samplesPerFrame = &temp0;
00134     if (!sampleRate)
00135         sampleRate      = &temp1;
00136 
00137     //*isMono = ((header >>  6) & 0x03) == 0x03;
00138 
00139     *sampleRate      = sSampleRates[sampleRateID] >> mode;
00140     bitRate          = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
00141     *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
00142     //av_log(NULL, AV_LOG_DEBUG,
00143     //       "sr:%d br:%d spf:%d l:%d m:%d\n",
00144     //       *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
00145 
00146     return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
00147 }
00148 
00149 static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
00150                                 int buf_size, void *data)
00151 {
00152     Mp3AudioContext *s = avctx->priv_data;
00153     int len;
00154     int lame_result;
00155 
00156     /* lame 3.91 dies on '1-channel interleaved' data */
00157 
00158     if (data) {
00159         if (s->stereo) {
00160             lame_result = lame_encode_buffer_interleaved(s->gfp, data,
00161                                                          avctx->frame_size,
00162                                                          s->buffer + s->buffer_index,
00163                                                          BUFFER_SIZE - s->buffer_index);
00164         } else {
00165             lame_result = lame_encode_buffer(s->gfp, data, data,
00166                                              avctx->frame_size, s->buffer +
00167                                              s->buffer_index, BUFFER_SIZE -
00168                                              s->buffer_index);
00169         }
00170     } else {
00171         lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
00172                                         BUFFER_SIZE - s->buffer_index);
00173     }
00174 
00175     if (lame_result < 0) {
00176         if (lame_result == -1) {
00177             /* output buffer too small */
00178             av_log(avctx, AV_LOG_ERROR,
00179                    "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
00180                    s->buffer_index, BUFFER_SIZE - s->buffer_index);
00181         }
00182         return -1;
00183     }
00184 
00185     s->buffer_index += lame_result;
00186 
00187     if (s->buffer_index < 4)
00188         return 0;
00189 
00190     len = mp3len(s->buffer, NULL, NULL);
00191     //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n",
00192     //       avctx->frame_size, len, s->buffer_index);
00193     if (len <= s->buffer_index) {
00194         memcpy(frame, s->buffer, len);
00195         s->buffer_index -= len;
00196 
00197         memmove(s->buffer, s->buffer + len, s->buffer_index);
00198         // FIXME fix the audio codec API, so we do not need the memcpy()
00199         /*for(i=0; i<len; i++) {
00200             av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
00201         }*/
00202         return len;
00203     } else
00204         return 0;
00205 }
00206 
00207 static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
00208 {
00209     Mp3AudioContext *s = avctx->priv_data;
00210 
00211     av_freep(&avctx->coded_frame);
00212 
00213     lame_close(s->gfp);
00214     return 0;
00215 }
00216 
00217 #define OFFSET(x) offsetof(Mp3AudioContext, x)
00218 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
00219 static const AVOption options[] = {
00220     { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { 1 }, 0, 1, AE },
00221     { NULL },
00222 };
00223 
00224 static const AVClass libmp3lame_class = {
00225     .class_name = "libmp3lame encoder",
00226     .item_name  = av_default_item_name,
00227     .option     = options,
00228     .version    = LIBAVUTIL_VERSION_INT,
00229 };
00230 
00231 AVCodec ff_libmp3lame_encoder = {
00232     .name                  = "libmp3lame",
00233     .type                  = AVMEDIA_TYPE_AUDIO,
00234     .id                    = CODEC_ID_MP3,
00235     .priv_data_size        = sizeof(Mp3AudioContext),
00236     .init                  = MP3lame_encode_init,
00237     .encode                = MP3lame_encode_frame,
00238     .close                 = MP3lame_encode_close,
00239     .capabilities          = CODEC_CAP_DELAY,
00240     .sample_fmts           = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
00241                                                              AV_SAMPLE_FMT_NONE },
00242     .supported_samplerates = sSampleRates,
00243     .long_name             = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
00244     .priv_class            = &libmp3lame_class,
00245 };
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