/* * Copyright (c) 2002 Mark Hills <mark@pogo.org.uk> * * 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 */ #include <vorbis/vorbisenc.h> #include "libavutil/avassert.h" #include "libavutil/fifo.h" #include "libavutil/opt.h" #include "avcodec.h" #include "audio_frame_queue.h" #include "internal.h" #include "vorbis.h" #include "vorbis_parser.h" /* Number of samples the user should send in each call. * This value is used because it is the LCD of all possible frame sizes, so * an output packet will always start at the same point as one of the input * packets. */ #define LIBVORBIS_FRAME_SIZE 64 #define BUFFER_SIZE (1024 * 64) typedef struct LibvorbisEncContext { AVClass *av_class; /**< class for AVOptions */ vorbis_info vi; /**< vorbis_info used during init */ vorbis_dsp_state vd; /**< DSP state used for analysis */ vorbis_block vb; /**< vorbis_block used for analysis */ AVFifoBuffer *pkt_fifo; /**< output packet buffer */ int eof; /**< end-of-file flag */ int dsp_initialized; /**< vd has been initialized */ vorbis_comment vc; /**< VorbisComment info */ double iblock; /**< impulse block bias option */ VorbisParseContext vp; /**< parse context to get durations */ AudioFrameQueue afq; /**< frame queue for timestamps */ } LibvorbisEncContext; static const AVOption options[] = { { "iblock", "Sets the impulse block bias", offsetof(LibvorbisEncContext, iblock), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -15, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM }, { NULL } }; static const AVCodecDefault defaults[] = { { "b", "0" }, { NULL }, }; static const AVClass vorbis_class = { .class_name = "libvorbis", .item_name = av_default_item_name, .option = options, .version = LIBAVUTIL_VERSION_INT, }; static int vorbis_error_to_averror(int ov_err) { switch (ov_err) { case OV_EFAULT: return AVERROR_BUG; case OV_EINVAL: return AVERROR(EINVAL); case OV_EIMPL: return AVERROR(EINVAL); default: return AVERROR_UNKNOWN; } } static av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx) { LibvorbisEncContext *s = avctx->priv_data; double cfreq; int ret; if (avctx->flags & CODEC_FLAG_QSCALE || !avctx->bit_rate) { /* variable bitrate * NOTE: we use the oggenc range of -1 to 10 for global_quality for * user convenience, but libvorbis uses -0.1 to 1.0. */ float q = avctx->global_quality / (float)FF_QP2LAMBDA; /* default to 3 if the user did not set quality or bitrate */ if (!(avctx->flags & CODEC_FLAG_QSCALE)) q = 3.0; if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels, avctx->sample_rate, q / 10.0))) goto error; } else { int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1; int maxrate = avctx->rc_max_rate > 0 ? avctx->rc_max_rate : -1; /* average bitrate */ if ((ret = vorbis_encode_setup_managed(vi, avctx->channels, avctx->sample_rate, maxrate, avctx->bit_rate, minrate))) goto error; /* variable bitrate by estimate, disable slow rate management */ if (minrate == -1 && maxrate == -1) if ((ret = vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL))) goto error; /* should not happen */ } /* cutoff frequency */ if (avctx->cutoff > 0) { cfreq = avctx->cutoff / 1000.0; if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq))) goto error; /* should not happen */ } /* impulse block bias */ if (s->iblock) { if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock))) goto error; } if (avctx->channels == 3 && avctx->channel_layout != (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER) || avctx->channels == 4 && avctx->channel_layout != AV_CH_LAYOUT_2_2 && avctx->channel_layout != AV_CH_LAYOUT_QUAD || avctx->channels == 5 && avctx->channel_layout != AV_CH_LAYOUT_5POINT0 && avctx->channel_layout != AV_CH_LAYOUT_5POINT0_BACK || avctx->channels == 6 && avctx->channel_layout != AV_CH_LAYOUT_5POINT1 && avctx->channel_layout != AV_CH_LAYOUT_5POINT1_BACK || avctx->channels == 7 && avctx->channel_layout != (AV_CH_LAYOUT_5POINT1|AV_CH_BACK_CENTER) || avctx->channels == 8 && avctx->channel_layout != AV_CH_LAYOUT_7POINT1) { if (avctx->channel_layout) { char name[32]; av_get_channel_layout_string(name, sizeof(name), avctx->channels, avctx->channel_layout); av_log(avctx, AV_LOG_ERROR, "%s not supported by Vorbis: " "output stream will have incorrect " "channel layout.\n", name); } else { av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The encoder " "will use Vorbis channel layout for " "%d channels.\n", avctx->channels); } } if ((ret = vorbis_encode_setup_init(vi))) goto error; return 0; error: return vorbis_error_to_averror(ret); } /* How many bytes are needed for a buffer of length 'l' */ static int xiph_len(int l) { return 1 + l / 255 + l; } static av_cold int libvorbis_encode_close(AVCodecContext *avctx) { LibvorbisEncContext *s = avctx->priv_data; /* notify vorbisenc this is EOF */ if (s->dsp_initialized) vorbis_analysis_wrote(&s->vd, 0); vorbis_block_clear(&s->vb); vorbis_dsp_clear(&s->vd); vorbis_info_clear(&s->vi); av_fifo_free(s->pkt_fifo); ff_af_queue_close(&s->afq); av_freep(&avctx->extradata); return 0; } static av_cold int libvorbis_encode_init(AVCodecContext *avctx) { LibvorbisEncContext *s = avctx->priv_data; ogg_packet header, header_comm, header_code; uint8_t *p; unsigned int offset; int ret; vorbis_info_init(&s->vi); if ((ret = libvorbis_setup(&s->vi, avctx))) { av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n"); goto error; } if ((ret = vorbis_analysis_init(&s->vd, &s->vi))) { av_log(avctx, AV_LOG_ERROR, "analysis init failed\n"); ret = vorbis_error_to_averror(ret); goto error; } s->dsp_initialized = 1; if ((ret = vorbis_block_init(&s->vd, &s->vb))) { av_log(avctx, AV_LOG_ERROR, "dsp init failed\n"); ret = vorbis_error_to_averror(ret); goto error; } vorbis_comment_init(&s->vc); if (!(avctx->flags & CODEC_FLAG_BITEXACT)) vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT); if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm, &header_code))) { ret = vorbis_error_to_averror(ret); goto error; } avctx->extradata_size = 1 + xiph_len(header.bytes) + xiph_len(header_comm.bytes) + header_code.bytes; p = avctx->extradata = av_malloc(avctx->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE); if (!p) { ret = AVERROR(ENOMEM); goto error; } p[0] = 2; offset = 1; offset += av_xiphlacing(&p[offset], header.bytes); offset += av_xiphlacing(&p[offset], header_comm.bytes); memcpy(&p[offset], header.packet, header.bytes); offset += header.bytes; memcpy(&p[offset], header_comm.packet, header_comm.bytes); offset += header_comm.bytes; memcpy(&p[offset], header_code.packet, header_code.bytes); offset += header_code.bytes; av_assert0(offset == avctx->extradata_size); if ((ret = avpriv_vorbis_parse_extradata(avctx, &s->vp)) < 0) { av_log(avctx, AV_LOG_ERROR, "invalid extradata\n"); return ret; } vorbis_comment_clear(&s->vc); avctx->frame_size = LIBVORBIS_FRAME_SIZE; ff_af_queue_init(avctx, &s->afq); s->pkt_fifo = av_fifo_alloc(BUFFER_SIZE); if (!s->pkt_fifo) { ret = AVERROR(ENOMEM); goto error; } return 0; error: libvorbis_encode_close(avctx); return ret; } static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr) { LibvorbisEncContext *s = avctx->priv_data; ogg_packet op; int ret, duration; /* send samples to libvorbis */ if (frame) { const int samples = frame->nb_samples; float **buffer; int c, channels = s->vi.channels; buffer = vorbis_analysis_buffer(&s->vd, samples); for (c = 0; c < channels; c++) { int co = (channels > 8) ? c : ff_vorbis_encoding_channel_layout_offsets[channels - 1][c]; memcpy(buffer[c], frame->extended_data[co], samples * sizeof(*buffer[c])); } if ((ret = vorbis_analysis_wrote(&s->vd, samples)) < 0) { av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n"); return vorbis_error_to_averror(ret); } if ((ret = ff_af_queue_add(&s->afq, frame)) < 0) return ret; } else { if (!s->eof) if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) { av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n"); return vorbis_error_to_averror(ret); } s->eof = 1; } /* retrieve available packets from libvorbis */ while ((ret = vorbis_analysis_blockout(&s->vd, &s->vb)) == 1) { if ((ret = vorbis_analysis(&s->vb, NULL)) < 0) break; if ((ret = vorbis_bitrate_addblock(&s->vb)) < 0) break; /* add any available packets to the output packet buffer */ while ((ret = vorbis_bitrate_flushpacket(&s->vd, &op)) == 1) { if (av_fifo_space(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) { av_log(avctx, AV_LOG_ERROR, "packet buffer is too small\n"); return AVERROR_BUG; } av_fifo_generic_write(s->pkt_fifo, &op, sizeof(ogg_packet), NULL); av_fifo_generic_write(s->pkt_fifo, op.packet, op.bytes, NULL); } if (ret < 0) { av_log(avctx, AV_LOG_ERROR, "error getting available packets\n"); break; } } if (ret < 0) { av_log(avctx, AV_LOG_ERROR, "error getting available packets\n"); return vorbis_error_to_averror(ret); } /* check for available packets */ if (av_fifo_size(s->pkt_fifo) < sizeof(ogg_packet)) return 0; av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL); if ((ret = ff_alloc_packet2(avctx, avpkt, op.bytes)) < 0) return ret; av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL); avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos); duration = avpriv_vorbis_parse_frame(&s->vp, avpkt->data, avpkt->size); if (duration > 0) { /* we do not know encoder delay until we get the first packet from * libvorbis, so we have to update the AudioFrameQueue counts */ if (!avctx->delay && s->afq.frames) { avctx->delay = duration; av_assert0(!s->afq.remaining_delay); s->afq.frames->duration += duration; s->afq.frames->pts -= duration; s->afq.remaining_samples += duration; } ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration); } *got_packet_ptr = 1; return 0; } AVCodec ff_libvorbis_encoder = { .name = "libvorbis", .long_name = NULL_IF_CONFIG_SMALL("libvorbis"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_VORBIS, .priv_data_size = sizeof(LibvorbisEncContext), .init = libvorbis_encode_init, .encode2 = libvorbis_encode_frame, .close = libvorbis_encode_close, .capabilities = CODEC_CAP_DELAY, .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, .priv_class = &vorbis_class, .defaults = defaults, };