libavcodec/libvorbis.c

   1 /*
   2  * copyright (c) 2002 Mark Hills <mark@pogo.org.uk>
   3  *
   4  * This file is part of Libav.
   5  *
   6  * Libav is free software; you can redistribute it and/or
   7  * modify it under the terms of the GNU Lesser General Public
   8  * License as published by the Free Software Foundation; either
   9  * version 2.1 of the License, or (at your option) any later version.
  10  *
  11  * Libav is distributed in the hope that it will be useful,
  12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  14  * Lesser General Public License for more details.
  15  *
  16  * You should have received a copy of the GNU Lesser General Public
  17  * License along with Libav; if not, write to the Free Software
  18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  19  */
  20
  21 /**
  22  * @file
  23  * Vorbis encoding support via libvorbisenc.
  24  * @author Mark Hills <mark@pogo.org.uk>
  25  */
  26
  27 #include <vorbis/vorbisenc.h>
  28
  29 #include "libavutil/fifo.h"
  30 #include "libavutil/opt.h"
  31 #include "avcodec.h"
  32 #include "audio_frame_queue.h"
  33 #include "bytestream.h"
  34 #include "internal.h"
  35 #include "vorbis.h"
  36 #include "vorbis_parser.h"
  37
  38 #undef NDEBUG
  39 #include <assert.h>
  40
  41 /* Number of samples the user should send in each call.
  42  * This value is used because it is the LCD of all possible frame sizes, so
  43  * an output packet will always start at the same point as one of the input
  44  * packets.
  45  */
  46 #define OGGVORBIS_FRAME_SIZE 64
  47
  48 #define BUFFER_SIZE (1024 * 64)
  49
  50 typedef struct OggVorbisContext {
  51     AVClass *av_class;                  /**< class for AVOptions            */
  52     vorbis_info vi;                     /**< vorbis_info used during init   */
  53     vorbis_dsp_state vd;                /**< DSP state used for analysis    */
  54     vorbis_block vb;                    /**< vorbis_block used for analysis */
  55     AVFifoBuffer *pkt_fifo;             /**< output packet buffer           */
  56     int eof;                            /**< end-of-file flag               */
  57     int dsp_initialized;                /**< vd has been initialized        */
  58     vorbis_comment vc;                  /**< VorbisComment info             */
  59     ogg_packet op;                      /**< ogg packet                     */
  60     double iblock;                      /**< impulse block bias option      */
  61     VorbisParseContext vp;              /**< parse context to get durations */
  62     AudioFrameQueue afq;                /**< frame queue for timestamps     */
  63 } OggVorbisContext;
  64
  65 static const AVOption options[] = {
  66     { "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 },
  67     { NULL }
  68 };
  69
  70 static const AVCodecDefault defaults[] = {
  71     { "b",  "0" },
  72     { NULL },
  73 };
  74
  75 static const AVClass class = { "libvorbis", av_default_item_name, options, LIBAVUTIL_VERSION_INT };
  76
  77
  78 static int vorbis_error_to_averror(int ov_err)
  79 {
  80     switch (ov_err) {
  81     case OV_EFAULT: return AVERROR_BUG;
  82     case OV_EINVAL: return AVERROR(EINVAL);
  83     case OV_EIMPL:  return AVERROR(EINVAL);
  84     default:        return AVERROR_UNKNOWN;
  85     }
  86 }
  87
  88 static av_cold int oggvorbis_init_encoder(vorbis_info *vi,
  89                                           AVCodecContext *avctx)
  90 {
  91     OggVorbisContext *s = avctx->priv_data;
  92     double cfreq;
  93     int ret;
  94
  95     if (avctx->flags & CODEC_FLAG_QSCALE || !avctx->bit_rate) {
  96         /* variable bitrate
  97          * NOTE: we use the oggenc range of -1 to 10 for global_quality for
  98          *       user convenience, but libvorbis uses -0.1 to 1.0.
  99          */
 100         float q = avctx->global_quality / (float)FF_QP2LAMBDA;
 101         /* default to 3 if the user did not set quality or bitrate */
 102         if (!(avctx->flags & CODEC_FLAG_QSCALE))
 103             q = 3.0;
 104         if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels,
 105                                            avctx->sample_rate,
 106                                            q / 10.0)))
 107             goto error;
 108     } else {
 109         int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1;
 110         int maxrate = avctx->rc_max_rate > 0 ? avctx->rc_max_rate : -1;
 111
 112         /* average bitrate */
 113         if ((ret = vorbis_encode_setup_managed(vi, avctx->channels,
 114                                                avctx->sample_rate, maxrate,
 115                                                avctx->bit_rate, minrate)))
 116             goto error;
 117
 118         /* variable bitrate by estimate, disable slow rate management */
 119         if (minrate == -1 && maxrate == -1)
 120             if ((ret = vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL)))
 121                 goto error;
 122     }
 123
 124     /* cutoff frequency */
 125     if (avctx->cutoff > 0) {
 126         cfreq = avctx->cutoff / 1000.0;
 127         if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))
 128             goto error;
 129     }
 130
 131     /* impulse block bias */
 132     if (s->iblock) {
 133         if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock)))
 134             goto error;
 135     }
 136
 137     if ((ret = vorbis_encode_setup_init(vi)))
 138         goto error;
 139
 140     return 0;
 141 error:
 142     return vorbis_error_to_averror(ret);
 143 }
 144
 145 /* How many bytes are needed for a buffer of length 'l' */
 146 static int xiph_len(int l)
 147 {
 148     return 1 + l / 255 + l;
 149 }
 150
 151 static av_cold int oggvorbis_encode_close(AVCodecContext *avctx)
 152 {
 153     OggVorbisContext *s = avctx->priv_data;
 154
 155     /* notify vorbisenc this is EOF */
 156     if (s->dsp_initialized)
 157         vorbis_analysis_wrote(&s->vd, 0);
 158
 159     vorbis_block_clear(&s->vb);
 160     vorbis_dsp_clear(&s->vd);
 161     vorbis_info_clear(&s->vi);
 162
 163     av_fifo_free(s->pkt_fifo);
 164     ff_af_queue_close(&s->afq);
 165 #if FF_API_OLD_ENCODE_AUDIO
 166     av_freep(&avctx->coded_frame);
 167 #endif
 168     av_freep(&avctx->extradata);
 169
 170     return 0;
 171 }
 172
 173 static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
 174 {
 175     OggVorbisContext *s = avctx->priv_data;
 176     ogg_packet header, header_comm, header_code;
 177     uint8_t *p;
 178     unsigned int offset;
 179     int ret;
 180
 181     vorbis_info_init(&s->vi);
 182     if ((ret = oggvorbis_init_encoder(&s->vi, avctx))) {
 183         av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n");
 184         goto error;
 185     }
 186     if ((ret = vorbis_analysis_init(&s->vd, &s->vi))) {
 187         av_log(avctx, AV_LOG_ERROR, "analysis init failed\n");
 188         ret = vorbis_error_to_averror(ret);
 189         goto error;
 190     }
 191     s->dsp_initialized = 1;
 192     if ((ret = vorbis_block_init(&s->vd, &s->vb))) {
 193         av_log(avctx, AV_LOG_ERROR, "dsp init failed\n");
 194         ret = vorbis_error_to_averror(ret);
 195         goto error;
 196     }
 197
 198     vorbis_comment_init(&s->vc);
 199     vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT);
 200
 201     if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm,
 202                                          &header_code))) {
 203         ret = vorbis_error_to_averror(ret);
 204         goto error;
 205     }
 206
 207     avctx->extradata_size = 1 + xiph_len(header.bytes)      +
 208                                 xiph_len(header_comm.bytes) +
 209                                 header_code.bytes;
 210     p = avctx->extradata = av_malloc(avctx->extradata_size +
 211                                      FF_INPUT_BUFFER_PADDING_SIZE);
 212     if (!p) {
 213         ret = AVERROR(ENOMEM);
 214         goto error;
 215     }
 216     p[0]    = 2;
 217     offset  = 1;
 218     offset += av_xiphlacing(&p[offset], header.bytes);
 219     offset += av_xiphlacing(&p[offset], header_comm.bytes);
 220     memcpy(&p[offset], header.packet, header.bytes);
 221     offset += header.bytes;
 222     memcpy(&p[offset], header_comm.packet, header_comm.bytes);
 223     offset += header_comm.bytes;
 224     memcpy(&p[offset], header_code.packet, header_code.bytes);
 225     offset += header_code.bytes;
 226     assert(offset == avctx->extradata_size);
 227
 228     if ((ret = avpriv_vorbis_parse_extradata(avctx, &s->vp)) < 0) {
 229         av_log(avctx, AV_LOG_ERROR, "invalid extradata\n");
 230         return ret;
 231     }
 232
 233     vorbis_comment_clear(&s->vc);
 234
 235     avctx->frame_size = OGGVORBIS_FRAME_SIZE;
 236     ff_af_queue_init(avctx, &s->afq);
 237
 238     s->pkt_fifo = av_fifo_alloc(BUFFER_SIZE);
 239     if (!s->pkt_fifo) {
 240         ret = AVERROR(ENOMEM);
 241         goto error;
 242     }
 243
 244 #if FF_API_OLD_ENCODE_AUDIO
 245     avctx->coded_frame = avcodec_alloc_frame();
 246     if (!avctx->coded_frame) {
 247         ret = AVERROR(ENOMEM);
 248         goto error;
 249     }
 250 #endif
 251
 252     return 0;
 253 error:
 254     oggvorbis_encode_close(avctx);
 255     return ret;
 256 }
 257
 258 static int oggvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
 259                                   const AVFrame *frame, int *got_packet_ptr)
 260 {
 261     OggVorbisContext *s = avctx->priv_data;
 262     ogg_packet op;
 263     int ret, duration;
 264
 265     /* send samples to libvorbis */
 266     if (frame) {
 267         const int samples = frame->nb_samples;
 268         float **buffer;
 269         int c, channels = s->vi.channels;
 270
 271         buffer = vorbis_analysis_buffer(&s->vd, samples);
 272         for (c = 0; c < channels; c++) {
 273             int co = (channels > 8) ? c :
 274                      ff_vorbis_encoding_channel_layout_offsets[channels - 1][c];
 275             memcpy(buffer[c], frame->extended_data[co],
 276                    samples * sizeof(*buffer[c]));
 277         }
 278         if ((ret = vorbis_analysis_wrote(&s->vd, samples)) < 0) {
 279             av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
 280             return vorbis_error_to_averror(ret);
 281         }
 282         if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
 283             return ret;
 284     } else {
 285         if (!s->eof)
 286             if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) {
 287                 av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
 288                 return vorbis_error_to_averror(ret);
 289             }
 290         s->eof = 1;
 291     }
 292
 293     /* retrieve available packets from libvorbis */
 294     while ((ret = vorbis_analysis_blockout(&s->vd, &s->vb)) == 1) {
 295         if ((ret = vorbis_analysis(&s->vb, NULL)) < 0)
 296             break;
 297         if ((ret = vorbis_bitrate_addblock(&s->vb)) < 0)
 298             break;
 299
 300         /* add any available packets to the output packet buffer */
 301         while ((ret = vorbis_bitrate_flushpacket(&s->vd, &op)) == 1) {
 302             if (av_fifo_space(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) {
 303                 av_log(avctx, AV_LOG_ERROR, "packet buffer is too small");
 304                 return AVERROR_BUG;
 305             }
 306             av_fifo_generic_write(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
 307             av_fifo_generic_write(s->pkt_fifo, op.packet, op.bytes, NULL);
 308         }
 309         if (ret < 0) {
 310             av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
 311             break;
 312         }
 313     }
 314     if (ret < 0) {
 315         av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
 316         return vorbis_error_to_averror(ret);
 317     }
 318
 319     /* check for available packets */
 320     if (av_fifo_size(s->pkt_fifo) < sizeof(ogg_packet))
 321         return 0;
 322
 323     av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
 324
 325     if ((ret = ff_alloc_packet(avpkt, op.bytes))) {
 326         av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
 327         return ret;
 328     }
 329     av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL);
 330
 331     avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos);
 332
 333     duration = avpriv_vorbis_parse_frame(&s->vp, avpkt->data, avpkt->size);
 334     if (duration > 0) {
 335         /* we do not know encoder delay until we get the first packet from
 336          * libvorbis, so we have to update the AudioFrameQueue counts */
 337         if (!avctx->delay) {
 338             avctx->delay              = duration;
 339             s->afq.remaining_delay   += duration;
 340             s->afq.remaining_samples += duration;
 341         }
 342         ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);
 343     }
 344
 345     *got_packet_ptr = 1;
 346     return 0;
 347 }
 348
 349 AVCodec ff_libvorbis_encoder = {
 350     .name           = "libvorbis",
 351     .type           = AVMEDIA_TYPE_AUDIO,
 352     .id             = AV_CODEC_ID_VORBIS,
 353     .priv_data_size = sizeof(OggVorbisContext),
 354     .init           = oggvorbis_encode_init,
 355     .encode2        = oggvorbis_encode_frame,
 356     .close          = oggvorbis_encode_close,
 357     .capabilities   = CODEC_CAP_DELAY,
 358     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
 359                                                       AV_SAMPLE_FMT_NONE },
 360     .long_name      = NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),
 361     .priv_class     = &class,
 362     .defaults       = defaults,
 363 };