view libvorbis.c @ 12197:fbf4d5b1b664 libavcodec

Remove FF_MM_SSE2/3 flags for CPUs where this is generally not faster than regular MMX code. Examples of this are the Core1 CPU. Instead, set a new flag, FF_MM_SSE2/3SLOW, which can be checked for particular SSE2/3 functions that have been checked specifically on such CPUs and are actually faster than their MMX counterparts. In addition, use this flag to enable particular VP8 and LPC SSE2 functions that are faster than their MMX counterparts. Based on a patch by Loren Merritt <lorenm AT u washington edu>.
author rbultje
date Mon, 19 Jul 2010 22:38:23 +0000
parents 24649290a14f
children a2c993c7ae90
line wrap: on
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/*
 * 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;
            if (context->buffer_index + sizeof(ogg_packet) + op.bytes > BUFFER_SIZE) {
                av_log(avccontext, AV_LOG_ERROR, "libvorbis: buffer overflow.");
                return -1;
            }
            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

        if (l > buf_size) {
            av_log(avccontext, AV_LOG_ERROR, "libvorbis: buffer overflow.");
            return -1;
        }

        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"),
} ;