X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Flibvorbis.c;h=88da705a3297a1bbfb2c3e487b2a172c8a8073ec;hb=348493db60de19d1997fd2861e130720218b9fcf;hp=6772ee4b396582557ae2105048a5dc8df1a1cf38;hpb=e5a5ea9e894c23f6224fceaef89e105f2c672396;p=ffmpeg diff --git a/libavcodec/libvorbis.c b/libavcodec/libvorbis.c index 6772ee4b396..88da705a329 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 */ @@ -26,8 +26,10 @@ #include +#include "libavutil/opt.h" #include "avcodec.h" #include "bytestream.h" +#include "vorbis.h" #undef NDEBUG #include @@ -37,6 +39,7 @@ #define BUFFER_SIZE (1024*64) typedef struct OggVorbisContext { + AVClass *av_class; vorbis_info vi ; vorbis_dsp_state vd ; vorbis_block vb ; @@ -47,10 +50,18 @@ typedef struct OggVorbisContext { /* 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, {.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) { @@ -68,11 +79,10 @@ static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avcco avccontext->sample_rate, minrate, avccontext->bit_rate, maxrate)) return -1; -#ifdef OGGVORBIS_VBR_BY_ESTIMATE - /* variable bitrate by estimate */ - if(vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE_AVG, NULL)) - return -1; -#endif + /* 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 */ @@ -82,14 +92,21 @@ static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avcco 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, len; + unsigned int offset; vorbis_info_init(&context->vi) ; if(oggvorbis_init_encoder(&context->vi, avccontext) < 0) { @@ -105,9 +122,11 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext) { 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); + 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); @@ -118,8 +137,7 @@ 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); vorbis_dsp_clear(&context->vd); @@ -145,18 +163,16 @@ static int oggvorbis_encode_frame(AVCodecContext *avccontext, int l; if(data) { - int samples = OGGVORBIS_FRAME_SIZE; + const int samples = avccontext->frame_size; float **buffer ; + int c, channels = context->vi.channels; buffer = vorbis_analysis_buffer(&context->vd, samples) ; - if(context->vi.channels == 1) { + 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[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; - } + buffer[c][l]=audio[l*channels+co]/32768.f; } vorbis_analysis_wrote(&context->vd, samples) ; } else { @@ -172,8 +188,12 @@ static int oggvorbis_encode_frame(AVCodecContext *avccontext, 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); @@ -191,9 +211,14 @@ static int oggvorbis_encode_frame(AVCodecContext *avccontext, 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); - memcpy(context->buffer, context->buffer + l + sizeof(ogg_packet), context->buffer_index); + memmove(context->buffer, context->buffer + l + sizeof(ogg_packet), context->buffer_index); // av_log(avccontext, AV_LOG_DEBUG, "E%d\n", l); } @@ -218,7 +243,7 @@ static av_cold int oggvorbis_encode_close(AVCodecContext *avccontext) { } -AVCodec libvorbis_encoder = { +AVCodec ff_libvorbis_encoder = { "libvorbis", AVMEDIA_TYPE_AUDIO, CODEC_ID_VORBIS, @@ -227,6 +252,7 @@ AVCodec libvorbis_encoder = { oggvorbis_encode_frame, oggvorbis_encode_close, .capabilities= CODEC_CAP_DELAY, - .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, + .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, .long_name= NULL_IF_CONFIG_SMALL("libvorbis Vorbis"), + .priv_class= &class, } ;