X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Flibvorbis.c;h=25e600671f052aff4034fd747d9aeebc564c46c4;hb=bcc73960657538f601dc90076e30df3cc6032569;hp=a7044a23b5713530c147a23f89480fde623046e6;hpb=1204a13c48f9c5c5158bf1d6630d7cea8b20ea9c;p=ffmpeg diff --git a/libavcodec/libvorbis.c b/libavcodec/libvorbis.c index a7044a23b57..25e600671f0 100644 --- a/libavcodec/libvorbis.c +++ b/libavcodec/libvorbis.c @@ -1,20 +1,20 @@ /* * copyright (c) 2002 Mark Hills * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav 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, + * Libav 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 + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -26,89 +26,116 @@ #include +#include "libavutil/opt.h" #include "avcodec.h" #include "bytestream.h" +#include "vorbis.h" +#include "libavutil/mathematics.h" #undef NDEBUG #include #define OGGVORBIS_FRAME_SIZE 64 -#define BUFFER_SIZE (1024*64) +#define BUFFER_SIZE (1024 * 64) typedef struct OggVorbisContext { - vorbis_info vi ; - vorbis_dsp_state vd ; - vorbis_block vb ; + 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 ; + vorbis_comment vc; ogg_packet op; -} OggVorbisContext ; + double iblock; +} OggVorbisContext; -static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avccontext) { +static const AVOption options[] = { + { "iblock", "Sets the impulse block bias", offsetof(OggVorbisContext, iblock), AV_OPT_TYPE_DOUBLE, { .dbl = 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) { + 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)) + 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)) + 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)) + if (minrate == -1 && maxrate == -1) + if (vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL)) return -1; } /* cutoff frequency */ - if(avccontext->cutoff > 0) { + if (avccontext->cutoff > 0) { cfreq = avccontext->cutoff / 1000.0; - if(vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)) + 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); } -static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext) { - OggVorbisContext *context = avccontext->priv_data ; +/* 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, len; + 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_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_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_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; + &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); @@ -117,62 +144,64 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext) { 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); + assert(offset == avccontext->extradata_size); -/* vorbis_block_clear(&context->vb); +#if 0 + vorbis_block_clear(&context->vb); vorbis_dsp_clear(&context->vd); - vorbis_info_clear(&context->vi);*/ + vorbis_info_clear(&context->vi); +#endif vorbis_comment_clear(&context->vc); - avccontext->frame_size = OGGVORBIS_FRAME_SIZE ; + avccontext->frame_size = OGGVORBIS_FRAME_SIZE; - avccontext->coded_frame= avcodec_alloc_frame(); - avccontext->coded_frame->key_frame= 1; + avccontext->coded_frame = avcodec_alloc_frame(); + avccontext->coded_frame->key_frame = 1; - return 0 ; + return 0; } - static int oggvorbis_encode_frame(AVCodecContext *avccontext, unsigned char *packets, - int buf_size, void *data) + int buf_size, void *data) { - OggVorbisContext *context = avccontext->priv_data ; - ogg_packet op ; - signed short *audio = data ; + OggVorbisContext *context = avccontext->priv_data; + ogg_packet op; + signed short *audio = data; int l; - if(data) { - int samples = OGGVORBIS_FRAME_SIZE; - float **buffer ; - - 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; - } + 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) ; + vorbis_analysis_wrote(&context->vd, samples); } else { - if(!context->eof) - vorbis_analysis_wrote(&context->vd, 0) ; + if (!context->eof) + vorbis_analysis_wrote(&context->vd, 0); context->eof = 1; } - while(vorbis_analysis_blockout(&context->vd, &context->vb) == 1) { + while (vorbis_analysis_blockout(&context->vd, &context->vb) == 1) { vorbis_analysis(&context->vb, NULL); - vorbis_bitrate_addblock(&context->vb) ; + vorbis_bitrate_addblock(&context->vb); - while(vorbis_bitrate_flushpacket(&context->vd, &op)) { + 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) + 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); @@ -181,15 +210,20 @@ static int oggvorbis_encode_frame(AVCodecContext *avccontext, } } - l=0; - if(context->buffer_index){ - ogg_packet *op2= (ogg_packet*)context->buffer; + 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); + 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); @@ -199,12 +233,12 @@ static int oggvorbis_encode_frame(AVCodecContext *avccontext, return l; } - -static av_cold int oggvorbis_encode_close(AVCodecContext *avccontext) { - OggVorbisContext *context = avccontext->priv_data ; +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_analysis_wrote(&context->vd, 0); /* notify vorbisenc this is EOF */ vorbis_block_clear(&context->vb); vorbis_dsp_clear(&context->vd); @@ -213,19 +247,19 @@ static av_cold int oggvorbis_encode_close(AVCodecContext *avccontext) { av_freep(&avccontext->coded_frame); av_freep(&avccontext->extradata); - return 0 ; + 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"), -} ; +AVCodec ff_libvorbis_encoder = { + .name = "libvorbis", + .type = AVMEDIA_TYPE_AUDIO, + .id = CODEC_ID_VORBIS, + .priv_data_size = sizeof(OggVorbisContext), + .init = oggvorbis_encode_init, + .encode = oggvorbis_encode_frame, + .close = 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, +};