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ffmpeg / libavcodec / libvorbis.c @ 2912e87a

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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 Libav.
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 *
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 * Libav is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * 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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 * Libav is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with Libav; 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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/**
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 * @file
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 * Ogg Vorbis codec support via libvorbisenc.
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 * @author Mark Hills <mark@pogo.org.uk>
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 */
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#include <vorbis/vorbisenc.h>
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "bytestream.h"
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#include "vorbis.h"
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#undef NDEBUG
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#include <assert.h>
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#define OGGVORBIS_FRAME_SIZE 64
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#define BUFFER_SIZE (1024*64)
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typedef struct OggVorbisContext {
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    AVClass *av_class;
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    vorbis_info vi ;
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    vorbis_dsp_state vd ;
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    vorbis_block vb ;
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    uint8_t buffer[BUFFER_SIZE];
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    int buffer_index;
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    int eof;
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    /* decoder */
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    vorbis_comment vc ;
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    ogg_packet op;
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    double iblock;
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} OggVorbisContext ;
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static const AVOption options[]={
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{"iblock", "Sets the impulse block bias", offsetof(OggVorbisContext, iblock), FF_OPT_TYPE_DOUBLE, 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 AVClass class = { "libvorbis", av_default_item_name, options, LIBAVUTIL_VERSION_INT };
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static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avccontext) {
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    OggVorbisContext *context = avccontext->priv_data ;
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    double cfreq;
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    if(avccontext->flags & CODEC_FLAG_QSCALE) {
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        /* variable bitrate */
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        if(vorbis_encode_setup_vbr(vi, avccontext->channels,
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                avccontext->sample_rate,
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                avccontext->global_quality / (float)FF_QP2LAMBDA / 10.0))
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            return -1;
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    } else {
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        int minrate = avccontext->rc_min_rate > 0 ? avccontext->rc_min_rate : -1;
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        int maxrate = avccontext->rc_min_rate > 0 ? avccontext->rc_max_rate : -1;
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        /* constant bitrate */
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        if(vorbis_encode_setup_managed(vi, avccontext->channels,
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                avccontext->sample_rate, minrate, avccontext->bit_rate, maxrate))
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            return -1;
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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(vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL))
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                return -1;
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    }
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    /* cutoff frequency */
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    if(avccontext->cutoff > 0) {
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        cfreq = avccontext->cutoff / 1000.0;
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        if(vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq))
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            return -1;
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    }
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    if(context->iblock){
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        vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &context->iblock);
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    }
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    return vorbis_encode_setup_init(vi);
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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) { return (1 + l / 255 + l); }
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static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext) {
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    OggVorbisContext *context = avccontext->priv_data ;
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    ogg_packet header, header_comm, header_code;
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    uint8_t *p;
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    unsigned int offset;
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    vorbis_info_init(&context->vi) ;
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    if(oggvorbis_init_encoder(&context->vi, avccontext) < 0) {
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        av_log(avccontext, AV_LOG_ERROR, "oggvorbis_encode_init: init_encoder failed\n") ;
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        return -1 ;
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    }
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    vorbis_analysis_init(&context->vd, &context->vi) ;
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    vorbis_block_init(&context->vd, &context->vb) ;
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    vorbis_comment_init(&context->vc);
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    vorbis_comment_add_tag(&context->vc, "encoder", LIBAVCODEC_IDENT) ;
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    vorbis_analysis_headerout(&context->vd, &context->vc, &header,
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                                &header_comm, &header_code);
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    avccontext->extradata_size=
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        1 + xiph_len(header.bytes) + xiph_len(header_comm.bytes) +
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        header_code.bytes;
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    p = avccontext->extradata =
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      av_malloc(avccontext->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
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    p[0] = 2;
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    offset = 1;
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    offset += av_xiphlacing(&p[offset], header.bytes);
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    offset += av_xiphlacing(&p[offset], header_comm.bytes);
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    memcpy(&p[offset], header.packet, header.bytes);
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    offset += header.bytes;
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    memcpy(&p[offset], header_comm.packet, header_comm.bytes);
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    offset += header_comm.bytes;
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    memcpy(&p[offset], header_code.packet, header_code.bytes);
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    offset += header_code.bytes;
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    assert(offset == avccontext->extradata_size);
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/*    vorbis_block_clear(&context->vb);
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    vorbis_dsp_clear(&context->vd);
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    vorbis_info_clear(&context->vi);*/
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    vorbis_comment_clear(&context->vc);
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    avccontext->frame_size = OGGVORBIS_FRAME_SIZE ;
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    avccontext->coded_frame= avcodec_alloc_frame();
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    avccontext->coded_frame->key_frame= 1;
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    return 0 ;
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}
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static int oggvorbis_encode_frame(AVCodecContext *avccontext,
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                                  unsigned char *packets,
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                           int buf_size, void *data)
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{
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    OggVorbisContext *context = avccontext->priv_data ;
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    ogg_packet op ;
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    signed short *audio = data ;
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    int l;
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    if(data) {
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        const int samples = avccontext->frame_size;
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        float **buffer ;
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        int c, channels = context->vi.channels;
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        buffer = vorbis_analysis_buffer(&context->vd, samples) ;
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        for (c = 0; c < channels; c++) {
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            int co = (channels > 8) ? c :
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                ff_vorbis_encoding_channel_layout_offsets[channels-1][c];
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            for(l = 0 ; l < samples ; l++)
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                buffer[c][l]=audio[l*channels+co]/32768.f;
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        }
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        vorbis_analysis_wrote(&context->vd, samples) ;
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    } else {
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        if(!context->eof)
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            vorbis_analysis_wrote(&context->vd, 0) ;
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        context->eof = 1;
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    }
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    while(vorbis_analysis_blockout(&context->vd, &context->vb) == 1) {
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        vorbis_analysis(&context->vb, NULL);
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        vorbis_bitrate_addblock(&context->vb) ;
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        while(vorbis_bitrate_flushpacket(&context->vd, &op)) {
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            /* i'd love to say the following line is a hack, but sadly it's
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             * not, apparently the end of stream decision is in libogg. */
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            if(op.bytes==1 && op.e_o_s)
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                continue;
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            if (context->buffer_index + sizeof(ogg_packet) + op.bytes > BUFFER_SIZE) {
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                av_log(avccontext, AV_LOG_ERROR, "libvorbis: buffer overflow.");
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                return -1;
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            }
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            memcpy(context->buffer + context->buffer_index, &op, sizeof(ogg_packet));
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            context->buffer_index += sizeof(ogg_packet);
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            memcpy(context->buffer + context->buffer_index, op.packet, op.bytes);
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            context->buffer_index += op.bytes;
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//            av_log(avccontext, AV_LOG_DEBUG, "e%d / %d\n", context->buffer_index, op.bytes);
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        }
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    }
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    l=0;
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    if(context->buffer_index){
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        ogg_packet *op2= (ogg_packet*)context->buffer;
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        op2->packet = context->buffer + sizeof(ogg_packet);
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        l=  op2->bytes;
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        avccontext->coded_frame->pts= av_rescale_q(op2->granulepos, (AVRational){1, avccontext->sample_rate}, avccontext->time_base);
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        //FIXME we should reorder the user supplied pts and not assume that they are spaced by 1/sample_rate
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        if (l > buf_size) {
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            av_log(avccontext, AV_LOG_ERROR, "libvorbis: buffer overflow.");
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            return -1;
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        }
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        memcpy(packets, op2->packet, l);
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        context->buffer_index -= l + sizeof(ogg_packet);
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        memmove(context->buffer, context->buffer + l + sizeof(ogg_packet), context->buffer_index);
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//        av_log(avccontext, AV_LOG_DEBUG, "E%d\n", l);
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    }
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    return l;
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}
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static av_cold int oggvorbis_encode_close(AVCodecContext *avccontext) {
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    OggVorbisContext *context = avccontext->priv_data ;
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/*  ogg_packet op ; */
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    vorbis_analysis_wrote(&context->vd, 0) ; /* notify vorbisenc this is EOF */
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    vorbis_block_clear(&context->vb);
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    vorbis_dsp_clear(&context->vd);
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    vorbis_info_clear(&context->vi);
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    av_freep(&avccontext->coded_frame);
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    av_freep(&avccontext->extradata);
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    return 0 ;
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}
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AVCodec ff_libvorbis_encoder = {
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    "libvorbis",
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    AVMEDIA_TYPE_AUDIO,
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    CODEC_ID_VORBIS,
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    sizeof(OggVorbisContext),
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    oggvorbis_encode_init,
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    oggvorbis_encode_frame,
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    oggvorbis_encode_close,
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    .capabilities= CODEC_CAP_DELAY,
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    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
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    .long_name= NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),
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    .priv_class= &class,
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} ;