libvorbisenc.c 13.4 KB
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/*
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 * Copyright (c) 2002 Mark Hills <mark@pogo.org.uk>
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 *
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 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
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 * 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
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */

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#include <vorbis/vorbisenc.h>

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#include "libavutil/avassert.h"
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#include "libavutil/fifo.h"
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "audio_frame_queue.h"
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#include "internal.h"
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#include "vorbis.h"
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#include "vorbis_parser.h"
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/* 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.
 */
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#define LIBVORBIS_FRAME_SIZE 64
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#define BUFFER_SIZE (1024 * 64)
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typedef struct LibvorbisEncContext {
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    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 */
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    AVFifoBuffer *pkt_fifo;             /**< output packet buffer           */
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    int eof;                            /**< end-of-file flag               */
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    int dsp_initialized;                /**< vd has been initialized        */
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    vorbis_comment vc;                  /**< VorbisComment info             */
    double iblock;                      /**< impulse block bias option      */
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    VorbisParseContext vp;              /**< parse context to get durations */
    AudioFrameQueue afq;                /**< frame queue for timestamps     */
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} LibvorbisEncContext;
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static const AVOption options[] = {
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    { "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 },
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    { NULL }
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};
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static const AVCodecDefault defaults[] = {
    { "b",  "0" },
    { NULL },
};

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static const AVClass vorbis_class = {
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    .class_name = "libvorbis",
    .item_name  = av_default_item_name,
    .option     = options,
    .version    = LIBAVUTIL_VERSION_INT,
};
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static int vorbis_error_to_averror(int ov_err)
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{
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    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;
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    }
}

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static av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx)
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{
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    LibvorbisEncContext *s = avctx->priv_data;
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    double cfreq;
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    int ret;
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    if (avctx->flags & CODEC_FLAG_QSCALE || !avctx->bit_rate) {
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        /* variable bitrate
         * NOTE: we use the oggenc range of -1 to 10 for global_quality for
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         *       user convenience, but libvorbis uses -0.1 to 1.0.
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         */
        float q = avctx->global_quality / (float)FF_QP2LAMBDA;
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        /* default to 3 if the user did not set quality or bitrate */
        if (!(avctx->flags & CODEC_FLAG_QSCALE))
            q = 3.0;
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        if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels,
                                           avctx->sample_rate,
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                                           q / 10.0)))
            goto error;
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    } else {
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        int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1;
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        int maxrate = avctx->rc_max_rate > 0 ? avctx->rc_max_rate : -1;
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        /* average bitrate */
        if ((ret = vorbis_encode_setup_managed(vi, avctx->channels,
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                                               avctx->sample_rate, maxrate,
                                               avctx->bit_rate, minrate)))
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            goto error;
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        /* variable bitrate by estimate, disable slow rate management */
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        if (minrate == -1 && maxrate == -1)
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            if ((ret = vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL)))
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                goto error; /* should not happen */
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    }
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    /* cutoff frequency */
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    if (avctx->cutoff > 0) {
        cfreq = avctx->cutoff / 1000.0;
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        if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))
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            goto error; /* should not happen */
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    }
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    /* impulse block bias */
    if (s->iblock) {
        if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock)))
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            goto error;
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    }

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    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) {
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            char name[32];
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            av_get_channel_layout_string(name, sizeof(name), avctx->channels,
                                         avctx->channel_layout);
            av_log(avctx, AV_LOG_ERROR, "%s not supported by Vorbis: "
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                                             "output stream will have incorrect "
                                             "channel layout.\n", name);
        } else {
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            av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The encoder "
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                                               "will use Vorbis channel layout for "
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                                               "%d channels.\n", avctx->channels);
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        }
    }

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    if ((ret = vorbis_encode_setup_init(vi)))
        goto error;

    return 0;
error:
    return vorbis_error_to_averror(ret);
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}

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/* How many bytes are needed for a buffer of length 'l' */
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static int xiph_len(int l)
{
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    return 1 + l / 255 + l;
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}
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static av_cold int libvorbis_encode_close(AVCodecContext *avctx)
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{
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    LibvorbisEncContext *s = avctx->priv_data;
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    /* notify vorbisenc this is EOF */
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    if (s->dsp_initialized)
        vorbis_analysis_wrote(&s->vd, 0);
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    vorbis_block_clear(&s->vb);
    vorbis_dsp_clear(&s->vd);
    vorbis_info_clear(&s->vi);
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    av_fifo_freep(&s->pkt_fifo);
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    ff_af_queue_close(&s->afq);
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    av_freep(&avctx->extradata);
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    return 0;
}

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static av_cold int libvorbis_encode_init(AVCodecContext *avctx)
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{
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    LibvorbisEncContext *s = avctx->priv_data;
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    ogg_packet header, header_comm, header_code;
    uint8_t *p;
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    unsigned int offset;
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    int ret;
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    vorbis_info_init(&s->vi);
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    if ((ret = libvorbis_setup(&s->vi, avctx))) {
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        av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n");
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        goto error;
    }
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    if ((ret = vorbis_analysis_init(&s->vd, &s->vi))) {
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        av_log(avctx, AV_LOG_ERROR, "analysis init failed\n");
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        ret = vorbis_error_to_averror(ret);
        goto error;
    }
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    s->dsp_initialized = 1;
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    if ((ret = vorbis_block_init(&s->vd, &s->vb))) {
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        av_log(avctx, AV_LOG_ERROR, "dsp init failed\n");
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        ret = vorbis_error_to_averror(ret);
        goto error;
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    }

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    vorbis_comment_init(&s->vc);
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    if (!(avctx->flags & CODEC_FLAG_BITEXACT))
        vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT);
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    if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm,
                                         &header_code))) {
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        ret = vorbis_error_to_averror(ret);
        goto error;
    }
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    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);
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    if (!p) {
        ret = AVERROR(ENOMEM);
        goto error;
    }
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    p[0]    = 2;
    offset  = 1;
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    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;
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    av_assert0(offset == avctx->extradata_size);
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    if ((ret = avpriv_vorbis_parse_extradata(avctx, &s->vp)) < 0) {
        av_log(avctx, AV_LOG_ERROR, "invalid extradata\n");
        return ret;
    }

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    vorbis_comment_clear(&s->vc);
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    avctx->frame_size = LIBVORBIS_FRAME_SIZE;
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    ff_af_queue_init(avctx, &s->afq);
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    s->pkt_fifo = av_fifo_alloc(BUFFER_SIZE);
    if (!s->pkt_fifo) {
        ret = AVERROR(ENOMEM);
        goto error;
    }
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    return 0;
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error:
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    libvorbis_encode_close(avctx);
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    return ret;
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}

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static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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                                  const AVFrame *frame, int *got_packet_ptr)
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{
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    LibvorbisEncContext *s = avctx->priv_data;
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    ogg_packet op;
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    int ret, duration;
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    /* send samples to libvorbis */
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    if (frame) {
        const int samples = frame->nb_samples;
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        float **buffer;
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        int c, channels = s->vi.channels;
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        buffer = vorbis_analysis_buffer(&s->vd, samples);
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        for (c = 0; c < channels; c++) {
            int co = (channels > 8) ? c :
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                     ff_vorbis_encoding_channel_layout_offsets[channels - 1][c];
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            memcpy(buffer[c], frame->extended_data[co],
                   samples * sizeof(*buffer[c]));
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        }
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        if ((ret = vorbis_analysis_wrote(&s->vd, samples)) < 0) {
            av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
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            return vorbis_error_to_averror(ret);
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        }
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        if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
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            return ret;
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    } else {
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        if (!s->eof && s->afq.frame_alloc)
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            if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) {
                av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
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                return vorbis_error_to_averror(ret);
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            }
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        s->eof = 1;
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    }
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    /* retrieve available packets from libvorbis */
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    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;
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        /* add any available packets to the output packet buffer */
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        while ((ret = vorbis_bitrate_flushpacket(&s->vd, &op)) == 1) {
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            if (av_fifo_space(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) {
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                av_log(avctx, AV_LOG_ERROR, "packet buffer is too small\n");
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                return AVERROR_BUG;
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            }
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            av_fifo_generic_write(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
            av_fifo_generic_write(s->pkt_fifo, op.packet, op.bytes, NULL);
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        }
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        if (ret < 0) {
            av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
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            break;
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        }
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    }
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    if (ret < 0) {
        av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
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        return vorbis_error_to_averror(ret);
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    }
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    /* 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);

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    if ((ret = ff_alloc_packet2(avctx, avpkt, op.bytes)) < 0)
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        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 */
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        if (!avctx->delay && s->afq.frames) {
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            avctx->delay              = duration;
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            av_assert0(!s->afq.remaining_delay);
            s->afq.frames->duration  += duration;
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            if (s->afq.frames->pts != AV_NOPTS_VALUE)
                s->afq.frames->pts       -= duration;
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            s->afq.remaining_samples += duration;
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        }
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        ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);
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    }

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    *got_packet_ptr = 1;
    return 0;
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}

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AVCodec ff_libvorbis_encoder = {
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    .name           = "libvorbis",
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    .long_name      = NULL_IF_CONFIG_SMALL("libvorbis"),
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    .type           = AVMEDIA_TYPE_AUDIO,
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    .id             = AV_CODEC_ID_VORBIS,
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    .priv_data_size = sizeof(LibvorbisEncContext),
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    .init           = libvorbis_encode_init,
    .encode2        = libvorbis_encode_frame,
    .close          = libvorbis_encode_close,
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    .capabilities   = CODEC_CAP_DELAY,
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    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
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                                                      AV_SAMPLE_FMT_NONE },
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    .priv_class     = &vorbis_class,
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    .defaults       = defaults,
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};