/* * 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 "libavutil/opt.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 { AVClass *av_class; 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; double iblock; } OggVorbisContext ; static const AVOption options[]={ {"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}, {NULL} }; static const AVClass class = { "libvorbis", av_default_item_name, options, LIBAVUTIL_VERSION_INT }; static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avccontext) { OggVorbisContext *context = avccontext->priv_data ; 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; } if(context->iblock){ vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &context->iblock); } return vorbis_encode_setup_init(vi); } /* 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_init(AVCodecContext *avccontext) { OggVorbisContext *context = avccontext->priv_data ; ogg_packet header, header_comm, header_code; uint8_t *p; unsigned int offset; 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); avccontext->extradata_size= 1 + xiph_len(header.bytes) + xiph_len(header_comm.bytes) + header_code.bytes; p = avccontext->extradata = av_malloc(avccontext->extradata_size + FF_INPUT_BUFFER_PADDING_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; assert(offset == 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 AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, .long_name= NULL_IF_CONFIG_SMALL("libvorbis Vorbis"), .priv_class= &class, } ;