/* * 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" #include "libavutil/mathematics.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), 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 const char * error(int oggerr, int *averr) { switch (oggerr) { case OV_EFAULT: *averr = AVERROR(EFAULT); return "internal error"; case OV_EIMPL: *averr = AVERROR(EINVAL); return "not supported"; case OV_EINVAL: *averr = AVERROR(EINVAL); return "invalid request"; default: *averr = AVERROR(EINVAL); return "unknown error"; } } static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avccontext) { OggVorbisContext *context = avccontext->priv_data; double cfreq; int r; if (avccontext->flags & CODEC_FLAG_QSCALE) { /* variable bitrate */ float quality = avccontext->global_quality / (float)FF_QP2LAMBDA; r = vorbis_encode_setup_vbr(vi, avccontext->channels, avccontext->sample_rate, quality / 10.0); if (r) { av_log(avccontext, AV_LOG_ERROR, "Unable to set quality to %g: %s\n", quality, error(r, &r)); return r; } } 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 */ r = vorbis_encode_setup_managed(vi, avccontext->channels, avccontext->sample_rate, minrate, avccontext->bit_rate, maxrate); if (r) { av_log(avccontext, AV_LOG_ERROR, "Unable to set CBR to %d: %s\n", avccontext->bit_rate, error(r, &r)); return r; } /* variable bitrate by estimate, disable slow rate management */ if (minrate == -1 && maxrate == -1) if (vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL)) return AVERROR(EINVAL); /* should not happen */ } /* cutoff frequency */ if (avccontext->cutoff > 0) { cfreq = avccontext->cutoff / 1000.0; if (vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)) return AVERROR(EINVAL); /* should not happen */ } if (context->iblock) { vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &context->iblock); } if (avccontext->channels == 3 && avccontext->channel_layout != (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER) || avccontext->channels == 4 && avccontext->channel_layout != AV_CH_LAYOUT_2_2 && avccontext->channel_layout != AV_CH_LAYOUT_QUAD || avccontext->channels == 5 && avccontext->channel_layout != AV_CH_LAYOUT_5POINT0 && avccontext->channel_layout != AV_CH_LAYOUT_5POINT0_BACK || avccontext->channels == 6 && avccontext->channel_layout != AV_CH_LAYOUT_5POINT1 && avccontext->channel_layout != AV_CH_LAYOUT_5POINT1_BACK || avccontext->channels == 7 && avccontext->channel_layout != (AV_CH_LAYOUT_5POINT1|AV_CH_BACK_CENTER) || avccontext->channels == 8 && avccontext->channel_layout != AV_CH_LAYOUT_7POINT1) { if (avccontext->channel_layout) { char name[32]; av_get_channel_layout_string(name, sizeof(name), avccontext->channels, avccontext->channel_layout); av_log(avccontext, AV_LOG_ERROR, "%s not supported by Vorbis: " "output stream will have incorrect " "channel layout.\n", name); } else { av_log(avccontext, AV_LOG_WARNING, "No channel layout specified. The encoder " "will use Vorbis channel layout for " "%d channels.\n", avccontext->channels); } } 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; int r; vorbis_info_init(&context->vi); r = oggvorbis_init_encoder(&context->vi, avccontext); if (r < 0) { av_log(avccontext, AV_LOG_ERROR, "oggvorbis_encode_init failed\n"); return r; } 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); #if 0 vorbis_block_clear(&context->vb); vorbis_dsp_clear(&context->vd); vorbis_info_clear(&context->vi); #endif 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.\n"); return AVERROR(EINVAL); } 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.\n"); return AVERROR(EINVAL); } 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 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, };