/* * Interface to libmp3lame for mp3 encoding * Copyright (c) 2002 Lennert Buytenhek * * 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 #include "libavutil/intreadwrite.h" #include "libavutil/log.h" #include "libavutil/opt.h" #include "avcodec.h" #include "internal.h" #include "mpegaudio.h" #include "mpegaudiodecheader.h" #define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4) typedef struct LAMEContext { AVClass *class; lame_global_flags *gfp; uint8_t buffer[BUFFER_SIZE]; int buffer_index; int reservoir; } LAMEContext; static av_cold int mp3lame_encode_close(AVCodecContext *avctx) { LAMEContext *s = avctx->priv_data; av_freep(&avctx->coded_frame); lame_close(s->gfp); return 0; } static av_cold int mp3lame_encode_init(AVCodecContext *avctx) { LAMEContext *s = avctx->priv_data; int ret; /* initialize LAME and get defaults */ if ((s->gfp = lame_init()) == NULL) return AVERROR(ENOMEM); /* channels */ if (avctx->channels > 2) { ret = AVERROR(EINVAL); goto error; } lame_set_num_channels(s->gfp, avctx->channels); lame_set_mode(s->gfp, avctx->channels > 1 ? JOINT_STEREO : MONO); /* sample rate */ lame_set_in_samplerate (s->gfp, avctx->sample_rate); lame_set_out_samplerate(s->gfp, avctx->sample_rate); /* algorithmic quality */ if (avctx->compression_level == FF_COMPRESSION_DEFAULT) lame_set_quality(s->gfp, 5); else lame_set_quality(s->gfp, avctx->compression_level); /* rate control */ if (avctx->flags & CODEC_FLAG_QSCALE) { lame_set_VBR(s->gfp, vbr_default); lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA); } else { if (avctx->bit_rate) lame_set_brate(s->gfp, avctx->bit_rate / 1000); } /* do not get a Xing VBR header frame from LAME */ lame_set_bWriteVbrTag(s->gfp,0); /* bit reservoir usage */ lame_set_disable_reservoir(s->gfp, !s->reservoir); /* set specified parameters */ if (lame_init_params(s->gfp) < 0) { ret = -1; goto error; } avctx->frame_size = lame_get_framesize(s->gfp); avctx->coded_frame = avcodec_alloc_frame(); if (!avctx->coded_frame) { ret = AVERROR(ENOMEM); goto error; } return 0; error: mp3lame_encode_close(avctx); return ret; } static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame, int buf_size, void *data) { LAMEContext *s = avctx->priv_data; MPADecodeHeader hdr; int len; int lame_result; if (data) { if (avctx->channels > 1) { 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) { 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; /* Move 1 frame from the LAME buffer to the output packet, if available. We have to parse the first frame header in the output buffer to determine the frame size. */ if (s->buffer_index < 4) return 0; if (avpriv_mpegaudio_decode_header(&hdr, AV_RB32(s->buffer))) { av_log(avctx, AV_LOG_ERROR, "free format output not supported\n"); return -1; } len = hdr.frame_size; av_dlog(avctx, "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); return len; } else return 0; } #define OFFSET(x) offsetof(LAMEContext, 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, }; static const AVCodecDefault libmp3lame_defaults[] = { { "b", "0" }, { NULL }, }; static const int libmp3lame_sample_rates[] = { 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0 }; AVCodec ff_libmp3lame_encoder = { .name = "libmp3lame", .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_MP3, .priv_data_size = sizeof(LAMEContext), .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 = libmp3lame_sample_rates, .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"), .priv_class = &libmp3lame_class, .defaults = libmp3lame_defaults, };