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/*

 * Copyright (c) 2002 Mark Hills 

 *

 * 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 

#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 OGGVORBIS_FRAME_SIZE 64

#define BUFFER_SIZE (1024 * 64)

typedef struct OggVorbisEncContext {

    AVClass *av_class;                  /**< class for AVOptions            */

    AVFrame frame;

    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     */

} OggVorbisEncContext;

static const AVOption options[] = {

    { "iblock", "Sets the impulse block bias", offsetof(OggVorbisEncContext, 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 oggvorbis_init_encoder(vorbis_info *vi,

                                          AVCodecContext *avctx)

{

    OggVorbisEncContext *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 oggvorbis_encode_close(AVCodecContext *avctx)

{

    OggVorbisEncContext *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 oggvorbis_encode_init(AVCodecContext *avctx)

{

    OggVorbisEncContext *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 = oggvorbis_init_encoder(&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 = OGGVORBIS_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:

    oggvorbis_encode_close(avctx);

    return ret;

}

static int oggvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,

                                  const AVFrame *frame, int *got_packet_ptr)

{

    OggVorbisEncContext *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(OggVorbisEncContext),

    .init           = oggvorbis_encode_init,

    .encode2        = oggvorbis_encode_frame,

    .close          = oggvorbis_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,

};