/* * Interface to libmp3lame for mp3 encoding * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org> * * This file is part of Libav. * * Libav 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. * * Libav 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 Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * Interface to libmp3lame for mp3 encoding. */ #include "libavutil/intreadwrite.h" #include "libavutil/log.h" #include "libavutil/opt.h" #include "avcodec.h" #include "mpegaudio.h" #include <lame/lame.h> #define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4) typedef struct Mp3AudioContext { AVClass *class; lame_global_flags *gfp; int stereo; uint8_t buffer[BUFFER_SIZE]; int buffer_index; int reservoir; } 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_set_mode(s->gfp, s->stereo ? JOINT_STEREO : MONO); 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_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA); } lame_set_bWriteVbrTag(s->gfp,0); lame_set_disable_reservoir(s->gfp, !s->reservoir); 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[] = { 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0 }; 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; i<len; i++) { av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]); }*/ return len; } else return 0; } static av_cold int MP3lame_encode_close(AVCodecContext *avctx) { Mp3AudioContext *s = avctx->priv_data; av_freep(&avctx->coded_frame); lame_close(s->gfp); return 0; } #define OFFSET(x) offsetof(Mp3AudioContext, x) #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM static const AVOption options[] = { { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { 1 }, 0, 1, AE }, { NULL }, }; static const AVClass libmp3lame_class = { .class_name = "libmp3lame encoder", .item_name = av_default_item_name, .option = options, .version = LIBAVUTIL_VERSION_INT, }; AVCodec ff_libmp3lame_encoder = { .name = "libmp3lame", .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_MP3, .priv_data_size = sizeof(Mp3AudioContext), .init = MP3lame_encode_init, .encode = MP3lame_encode_frame, .close = MP3lame_encode_close, .capabilities = CODEC_CAP_DELAY, .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, .supported_samplerates = sSampleRates, .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"), .priv_class = &libmp3lame_class, };