/* * Interface to libmp3lame for mp3 encoding * Copyright (c) 2002 Lennert Buytenhek * * 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 * Interface to libmp3lame for mp3 encoding. */ #include #include "libavutil/channel_layout.h" #include "libavutil/common.h" #include "libavutil/float_dsp.h" #include "libavutil/intreadwrite.h" #include "libavutil/log.h" #include "libavutil/opt.h" #include "avcodec.h" #include "audio_frame_queue.h" #include "internal.h" #include "mpegaudio.h" #include "mpegaudiodecheader.h" #define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4+1000) // FIXME: Buffer size to small? Adding 1000 to make up for it. typedef struct LAMEContext { AVClass *class; AVCodecContext *avctx; lame_global_flags *gfp; uint8_t *buffer; int buffer_index; int buffer_size; int reservoir; int joint_stereo; float *samples_flt[2]; AudioFrameQueue afq; AVFloatDSPContext fdsp; } LAMEContext; static int realloc_buffer(LAMEContext *s) { if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) { uint8_t *tmp; int new_size = s->buffer_index + 2 * BUFFER_SIZE; av_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size, new_size); tmp = av_realloc(s->buffer, new_size); if (!tmp) { av_freep(&s->buffer); s->buffer_size = s->buffer_index = 0; return AVERROR(ENOMEM); } s->buffer = tmp; s->buffer_size = new_size; } return 0; } static av_cold int mp3lame_encode_close(AVCodecContext *avctx) { LAMEContext *s = avctx->priv_data; av_freep(&s->samples_flt[0]); av_freep(&s->samples_flt[1]); av_freep(&s->buffer); ff_af_queue_close(&s->afq); lame_close(s->gfp); return 0; } static av_cold int mp3lame_encode_init(AVCodecContext *avctx) { LAMEContext *s = avctx->priv_data; int ret; s->avctx = avctx; /* initialize LAME and get defaults */ if ((s->gfp = lame_init()) == NULL) return AVERROR(ENOMEM); lame_set_num_channels(s->gfp, avctx->channels); lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : 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; } /* get encoder delay */ avctx->delay = lame_get_encoder_delay(s->gfp) + 528 + 1; ff_af_queue_init(avctx, &s->afq); avctx->frame_size = lame_get_framesize(s->gfp); /* allocate float sample buffers */ if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) { int ch; for (ch = 0; ch < avctx->channels; ch++) { s->samples_flt[ch] = av_malloc(avctx->frame_size * sizeof(*s->samples_flt[ch])); if (!s->samples_flt[ch]) { ret = AVERROR(ENOMEM); goto error; } } } ret = realloc_buffer(s); if (ret < 0) goto error; avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT); return 0; error: mp3lame_encode_close(avctx); return ret; } #define ENCODE_BUFFER(func, buf_type, buf_name) do { \ lame_result = func(s->gfp, \ (const buf_type *)buf_name[0], \ (const buf_type *)buf_name[1], frame->nb_samples, \ s->buffer + s->buffer_index, \ s->buffer_size - s->buffer_index); \ } while (0) static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr) { LAMEContext *s = avctx->priv_data; MPADecodeHeader hdr; int len, ret, ch; int lame_result; if (frame) { switch (avctx->sample_fmt) { case AV_SAMPLE_FMT_S16P: ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data); break; case AV_SAMPLE_FMT_S32P: ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data); break; case AV_SAMPLE_FMT_FLTP: if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) { av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n"); return AVERROR(EINVAL); } for (ch = 0; ch < avctx->channels; ch++) { s->fdsp.vector_fmul_scalar(s->samples_flt[ch], (const float *)frame->data[ch], 32768.0f, FFALIGN(frame->nb_samples, 8)); } ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt); break; default: return AVERROR_BUG; } } else { lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index, s->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, s->buffer_size - s->buffer_index); } return -1; } s->buffer_index += lame_result; ret = realloc_buffer(s); if (ret < 0) { av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n"); return ret; } /* add current frame to the queue */ if (frame) { if ((ret = ff_af_queue_add(&s->afq, frame)) < 0) return ret; } /* 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) { if ((ret = ff_alloc_packet2(avctx, avpkt, len)) < 0) return ret; memcpy(avpkt->data, s->buffer, len); s->buffer_index -= len; memmove(s->buffer, s->buffer + len, s->buffer_index); /* Get the next frame pts/duration */ ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts, &avpkt->duration); avpkt->size = len; *got_packet_ptr = 1; } 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, { .i64 = 1 }, 0, 1, AE }, { "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_INT, { .i64 = 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", .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_MP3, .priv_data_size = sizeof(LAMEContext), .init = mp3lame_encode_init, .encode2 = mp3lame_encode_frame, .close = mp3lame_encode_close, .capabilities = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME, .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE }, .supported_samplerates = libmp3lame_sample_rates, .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO, AV_CH_LAYOUT_STEREO, 0 }, .priv_class = &libmp3lame_class, .defaults = libmp3lame_defaults, };