PeerStreamer

/*

 * 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

 */

/**

 * @file

 * Ogg Vorbis codec support via libvorbisenc.

 * @author Mark Hills <mark@pogo.org.uk>

 */

#include <vorbis/vorbisenc.h>

#include "avcodec.h"

#include "bytestream.h"

#include "vorbis.h"

#undef NDEBUG

#include <assert.h>

#define OGGVORBIS_FRAME_SIZE 64

#define BUFFER_SIZE (1024*64)

typedef struct OggVorbisContext {

    vorbis_info vi ;

    vorbis_dsp_state vd ;

    vorbis_block vb ;

    uint8_t buffer[BUFFER_SIZE];

    int buffer_index;

    int eof;

    /* decoder */

    vorbis_comment vc ;

    ogg_packet op;

} OggVorbisContext ;

static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avccontext) {

    double cfreq;

    if(avccontext->flags & CODEC_FLAG_QSCALE) {

        /* variable bitrate */

        if(vorbis_encode_setup_vbr(vi, avccontext->channels,

                avccontext->sample_rate,

                avccontext->global_quality / (float)FF_QP2LAMBDA / 10.0))

            return -1;

    } else {

        int minrate = avccontext->rc_min_rate > 0 ? avccontext->rc_min_rate : -1;

        int maxrate = avccontext->rc_min_rate > 0 ? avccontext->rc_max_rate : -1;

        /* constant bitrate */

        if(vorbis_encode_setup_managed(vi, avccontext->channels,

                avccontext->sample_rate, minrate, avccontext->bit_rate, maxrate))

            return -1;

        /* variable bitrate by estimate, disable slow rate management */

        if(minrate == -1 && maxrate == -1)

            if(vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL))

                return -1;

    }

    /* cutoff frequency */

    if(avccontext->cutoff > 0) {

        cfreq = avccontext->cutoff / 1000.0;

        if(vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq))

            return -1;

    }

    return vorbis_encode_setup_init(vi);

}

static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext) {

    OggVorbisContext *context = avccontext->priv_data ;

    ogg_packet header, header_comm, header_code;

    uint8_t *p;

    unsigned int offset, len;

    vorbis_info_init(&context->vi) ;

    if(oggvorbis_init_encoder(&context->vi, avccontext) < 0) {

        av_log(avccontext, AV_LOG_ERROR, "oggvorbis_encode_init: init_encoder failed\n") ;

        return -1 ;

    }

    vorbis_analysis_init(&context->vd, &context->vi) ;

    vorbis_block_init(&context->vd, &context->vb) ;

    vorbis_comment_init(&context->vc);

    vorbis_comment_add_tag(&context->vc, "encoder", LIBAVCODEC_IDENT) ;

    vorbis_analysis_headerout(&context->vd, &context->vc, &header,

                                &header_comm, &header_code);

    len = header.bytes + header_comm.bytes +  header_code.bytes;

    avccontext->extradata_size= 64 + len + len/255;

    p = avccontext->extradata= av_mallocz(avccontext->extradata_size);

    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;

    avccontext->extradata_size = offset;

    avccontext->extradata= av_realloc(avccontext->extradata, avccontext->extradata_size);

/*    vorbis_block_clear(&context->vb);

    vorbis_dsp_clear(&context->vd);

    vorbis_info_clear(&context->vi);*/

    vorbis_comment_clear(&context->vc);

    avccontext->frame_size = OGGVORBIS_FRAME_SIZE ;

    avccontext->coded_frame= avcodec_alloc_frame();

    avccontext->coded_frame->key_frame= 1;

    return 0 ;

}

static int oggvorbis_encode_frame(AVCodecContext *avccontext,

                                  unsigned char *packets,

                           int buf_size, void *data)

{

    OggVorbisContext *context = avccontext->priv_data ;

    ogg_packet op ;

    signed short *audio = data ;

    int l;

    if(data) {

        const int samples = avccontext->frame_size;

        float **buffer ;

        int c, channels = context->vi.channels;

        buffer = vorbis_analysis_buffer(&context->vd, samples) ;

        for (c = 0; c < channels; c++) {

            int co = (channels > 8) ? c :

                ff_vorbis_encoding_channel_layout_offsets[channels-1][c];

            for(l = 0 ; l < samples ; l++)

                buffer[c][l]=audio[l*channels+co]/32768.f;

        }

        vorbis_analysis_wrote(&context->vd, samples) ;

    } else {

        if(!context->eof)

            vorbis_analysis_wrote(&context->vd, 0) ;

        context->eof = 1;

    }

    while(vorbis_analysis_blockout(&context->vd, &context->vb) == 1) {

        vorbis_analysis(&context->vb, NULL);

        vorbis_bitrate_addblock(&context->vb) ;

        while(vorbis_bitrate_flushpacket(&context->vd, &op)) {

            /* i'd love to say the following line is a hack, but sadly it's

             * not, apparently the end of stream decision is in libogg. */

            if(op.bytes==1 && op.e_o_s)

                continue;

            memcpy(context->buffer + context->buffer_index, &op, sizeof(ogg_packet));

            context->buffer_index += sizeof(ogg_packet);

            memcpy(context->buffer + context->buffer_index, op.packet, op.bytes);

            context->buffer_index += op.bytes;

//            av_log(avccontext, AV_LOG_DEBUG, "e%d / %d\n", context->buffer_index, op.bytes);

        }

    }

    l=0;

    if(context->buffer_index){

        ogg_packet *op2= (ogg_packet*)context->buffer;

        op2->packet = context->buffer + sizeof(ogg_packet);

        l=  op2->bytes;

        avccontext->coded_frame->pts= av_rescale_q(op2->granulepos, (AVRational){1, avccontext->sample_rate}, avccontext->time_base);

        //FIXME we should reorder the user supplied pts and not assume that they are spaced by 1/sample_rate

        memcpy(packets, op2->packet, l);

        context->buffer_index -= l + sizeof(ogg_packet);

        memmove(context->buffer, context->buffer + l + sizeof(ogg_packet), context->buffer_index);

//        av_log(avccontext, AV_LOG_DEBUG, "E%d\n", l);

    }

    return l;

}

static av_cold int oggvorbis_encode_close(AVCodecContext *avccontext) {

    OggVorbisContext *context = avccontext->priv_data ;

/*  ogg_packet op ; */

    vorbis_analysis_wrote(&context->vd, 0) ; /* notify vorbisenc this is EOF */

    vorbis_block_clear(&context->vb);

    vorbis_dsp_clear(&context->vd);

    vorbis_info_clear(&context->vi);

    av_freep(&avccontext->coded_frame);

    av_freep(&avccontext->extradata);

    return 0 ;

}

AVCodec libvorbis_encoder = {

    "libvorbis",

    AVMEDIA_TYPE_AUDIO,

    CODEC_ID_VORBIS,

    sizeof(OggVorbisContext),

    oggvorbis_encode_init,

    oggvorbis_encode_frame,

    oggvorbis_encode_close,

    .capabilities= CODEC_CAP_DELAY,

    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

    .long_name= NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),

} ;