Mercurial > libavcodec.hg
view libvorbis.c @ 5470:fea6732bc199 libavcodec
dnxhd 120 progressive support
author | bcoudurier |
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date | Sat, 04 Aug 2007 13:26:29 +0000 |
parents | 2b72f9bc4f06 |
children | c2ab2ac31edb |
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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 oggvorbis.c * Ogg Vorbis codec support via libvorbisenc. * @author Mark Hills <mark@pogo.org.uk> */ #include <vorbis/vorbisenc.h> #include "avcodec.h" #include "bytestream.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; /* decoder */ vorbis_comment vc ; ogg_packet op; } OggVorbisContext ; static 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)) return -1; } else { /* constant bitrate */ if(vorbis_encode_setup_managed(vi, avccontext->channels, avccontext->sample_rate, -1, avccontext->bit_rate, -1)) return -1; #ifdef OGGVORBIS_VBR_BY_ESTIMATE /* variable bitrate by estimate */ if(vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE_AVG, NULL)) return -1; #endif } /* 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 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") ; 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 ; float **buffer ; ogg_packet op ; signed short *audio = data ; int l, samples = data ? OGGVORBIS_FRAME_SIZE : 0; buffer = vorbis_analysis_buffer(&context->vd, samples) ; if(context->vi.channels == 1) { for(l = 0 ; l < samples ; l++) buffer[0][l]=audio[l]/32768.f; } else { for(l = 0 ; l < samples ; l++){ buffer[0][l]=audio[l*2]/32768.f; buffer[1][l]=audio[l*2+1]/32768.f; } } vorbis_analysis_wrote(&context->vd, samples) ; 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) 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); memcpy(context->buffer, context->buffer + l + sizeof(ogg_packet), context->buffer_index); // av_log(avccontext, AV_LOG_DEBUG, "E%d\n", l); } return l; } static 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", CODEC_TYPE_AUDIO, CODEC_ID_VORBIS, sizeof(OggVorbisContext), oggvorbis_encode_init, oggvorbis_encode_frame, oggvorbis_encode_close, .capabilities= CODEC_CAP_DELAY, } ; static int oggvorbis_decode_init(AVCodecContext *avccontext) { OggVorbisContext *context = avccontext->priv_data ; uint8_t *p= avccontext->extradata; int i, hsizes[3]; unsigned char *headers[3], *extradata = avccontext->extradata; vorbis_info_init(&context->vi) ; vorbis_comment_init(&context->vc) ; if(! avccontext->extradata_size || ! p) { av_log(avccontext, AV_LOG_ERROR, "vorbis extradata absent\n"); return -1; } if(p[0] == 0 && p[1] == 30) { for(i = 0; i < 3; i++){ hsizes[i] = bytestream_get_be16(&p); headers[i] = p; p += hsizes[i]; } } else if(*p == 2) { unsigned int offset = 1; p++; for(i=0; i<2; i++) { hsizes[i] = 0; while((*p == 0xFF) && (offset < avccontext->extradata_size)) { hsizes[i] += 0xFF; offset++; p++; } if(offset >= avccontext->extradata_size - 1) { av_log(avccontext, AV_LOG_ERROR, "vorbis header sizes damaged\n"); return -1; } hsizes[i] += *p; offset++; p++; } hsizes[2] = avccontext->extradata_size - hsizes[0]-hsizes[1]-offset; #if 0 av_log(avccontext, AV_LOG_DEBUG, "vorbis header sizes: %d, %d, %d, / extradata_len is %d \n", hsizes[0], hsizes[1], hsizes[2], avccontext->extradata_size); #endif headers[0] = extradata + offset; headers[1] = extradata + offset + hsizes[0]; headers[2] = extradata + offset + hsizes[0] + hsizes[1]; } else { av_log(avccontext, AV_LOG_ERROR, "vorbis initial header len is wrong: %d\n", *p); return -1; } for(i=0; i<3; i++){ context->op.b_o_s= i==0; context->op.bytes = hsizes[i]; context->op.packet = headers[i]; if(vorbis_synthesis_headerin(&context->vi, &context->vc, &context->op)<0){ av_log(avccontext, AV_LOG_ERROR, "%d. vorbis header damaged\n", i+1); return -1; } } avccontext->channels = context->vi.channels; avccontext->sample_rate = context->vi.rate; avccontext->time_base= (AVRational){1, avccontext->sample_rate}; vorbis_synthesis_init(&context->vd, &context->vi); vorbis_block_init(&context->vd, &context->vb); return 0 ; } static inline int conv(int samples, float **pcm, char *buf, int channels) { int i, j, val ; ogg_int16_t *ptr, *data = (ogg_int16_t*)buf ; float *mono ; for(i = 0 ; i < channels ; i++){ ptr = &data[i]; mono = pcm[i] ; for(j = 0 ; j < samples ; j++) { val = mono[j] * 32767.f; if(val > 32767) val = 32767 ; if(val < -32768) val = -32768 ; *ptr = val ; ptr += channels; } } return 0 ; } static int oggvorbis_decode_frame(AVCodecContext *avccontext, void *data, int *data_size, uint8_t *buf, int buf_size) { OggVorbisContext *context = avccontext->priv_data ; float **pcm ; ogg_packet *op= &context->op; int samples, total_samples, total_bytes; if(!buf_size){ //FIXME flush return 0; } op->packet = buf; op->bytes = buf_size; // av_log(avccontext, AV_LOG_DEBUG, "%d %d %d %"PRId64" %"PRId64" %d %d\n", op->bytes, op->b_o_s, op->e_o_s, op->granulepos, op->packetno, buf_size, context->vi.rate); /* for(i=0; i<op->bytes; i++) av_log(avccontext, AV_LOG_DEBUG, "%02X ", op->packet[i]); av_log(avccontext, AV_LOG_DEBUG, "\n");*/ if(vorbis_synthesis(&context->vb, op) == 0) vorbis_synthesis_blockin(&context->vd, &context->vb) ; total_samples = 0 ; total_bytes = 0 ; while((samples = vorbis_synthesis_pcmout(&context->vd, &pcm)) > 0) { conv(samples, pcm, (char*)data + total_bytes, context->vi.channels) ; total_bytes += samples * 2 * context->vi.channels ; total_samples += samples ; vorbis_synthesis_read(&context->vd, samples) ; } *data_size = total_bytes ; return buf_size ; } static int oggvorbis_decode_close(AVCodecContext *avccontext) { OggVorbisContext *context = avccontext->priv_data ; vorbis_info_clear(&context->vi) ; vorbis_comment_clear(&context->vc) ; return 0 ; } AVCodec libvorbis_decoder = { "libvorbis", CODEC_TYPE_AUDIO, CODEC_ID_VORBIS, sizeof(OggVorbisContext), oggvorbis_decode_init, NULL, oggvorbis_decode_close, oggvorbis_decode_frame, .capabilities= CODEC_CAP_DELAY, } ;