/*
* copyright (c) 2002 Mark Hills <mark@pogo.org.uk>
*
- * 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
*/
/**
- * @file oggvorbis.c
- * Ogg Vorbis codec support via libvorbisenc.
+ * @file
+ * Vorbis encoding support via libvorbisenc.
* @author Mark Hills <mark@pogo.org.uk>
*/
#include <vorbis/vorbisenc.h>
+#include "libavutil/fifo.h"
+#include "libavutil/opt.h"
#include "avcodec.h"
+#include "audio_frame_queue.h"
#include "bytestream.h"
+#include "internal.h"
+#include "vorbis.h"
+#include "vorbis_parser.h"
#undef NDEBUG
#include <assert.h>
+/* Number of samples the user should send in each call.
+ * This value is used because it is the LCD of all possible frame sizes, so
+ * an output packet will always start at the same point as one of the input
+ * packets.
+ */
#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 ;
- uint8_t buffer[BUFFER_SIZE];
- int buffer_index;
-
- /* decoder */
- vorbis_comment vc ;
- ogg_packet op;
-} OggVorbisContext ;
-
+ AVClass *av_class; /**< class for AVOptions */
+ vorbis_info vi; /**< vorbis_info used during init */
+ vorbis_dsp_state vd; /**< DSP state used for analysis */
+ vorbis_block vb; /**< vorbis_block used for analysis */
+ AVFifoBuffer *pkt_fifo; /**< output packet buffer */
+ int eof; /**< end-of-file flag */
+ int dsp_initialized; /**< vd has been initialized */
+ vorbis_comment vc; /**< VorbisComment info */
+ ogg_packet op; /**< ogg packet */
+ double iblock; /**< impulse block bias option */
+ VorbisParseContext vp; /**< parse context to get durations */
+ AudioFrameQueue afq; /**< frame queue for timestamps */
+} 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 AVCodecDefault defaults[] = {
+ { "b", "0" },
+ { NULL },
+};
+
+static const AVClass class = { "libvorbis", av_default_item_name, options, LIBAVUTIL_VERSION_INT };
+
+
+static int vorbis_error_to_averror(int ov_err)
+{
+ switch (ov_err) {
+ case OV_EFAULT: return AVERROR_BUG;
+ case OV_EINVAL: return AVERROR(EINVAL);
+ case OV_EIMPL: return AVERROR(EINVAL);
+ default: return AVERROR_UNKNOWN;
+ }
+}
-static int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avccontext) {
+static av_cold int oggvorbis_init_encoder(vorbis_info *vi,
+ AVCodecContext *avctx)
+{
+ OggVorbisContext *s = avctx->priv_data;
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;
+ int ret;
+
+ if (avctx->flags & CODEC_FLAG_QSCALE || !avctx->bit_rate) {
+ /* variable bitrate
+ * NOTE: we use the oggenc range of -1 to 10 for global_quality for
+ * user convenience, but libvorbis uses -0.1 to 1.0.
+ */
+ float q = avctx->global_quality / (float)FF_QP2LAMBDA;
+ /* default to 3 if the user did not set quality or bitrate */
+ if (!(avctx->flags & CODEC_FLAG_QSCALE))
+ q = 3.0;
+ if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels,
+ avctx->sample_rate,
+ q / 10.0)))
+ goto error;
} 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
+ int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1;
+ int maxrate = avctx->rc_max_rate > 0 ? avctx->rc_max_rate : -1;
+
+ /* average bitrate */
+ if ((ret = vorbis_encode_setup_managed(vi, avctx->channels,
+ avctx->sample_rate, maxrate,
+ avctx->bit_rate, minrate)))
+ goto error;
+
+ /* variable bitrate by estimate, disable slow rate management */
+ if (minrate == -1 && maxrate == -1)
+ if ((ret = vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL)))
+ goto error;
}
/* cutoff frequency */
- if(avccontext->cutoff > 0) {
- cfreq = avccontext->cutoff / 1000.0;
- if(vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq))
- return -1;
+ if (avctx->cutoff > 0) {
+ cfreq = avctx->cutoff / 1000.0;
+ if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))
+ goto error;
}
- return vorbis_encode_setup_init(vi);
+ /* impulse block bias */
+ if (s->iblock) {
+ if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock)))
+ goto error;
+ }
+
+ if ((ret = vorbis_encode_setup_init(vi)))
+ goto error;
+
+ return 0;
+error:
+ return vorbis_error_to_averror(ret);
+}
+
+/* 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_close(AVCodecContext *avctx)
+{
+ OggVorbisContext *s = avctx->priv_data;
+
+ /* notify vorbisenc this is EOF */
+ if (s->dsp_initialized)
+ vorbis_analysis_wrote(&s->vd, 0);
+
+ vorbis_block_clear(&s->vb);
+ vorbis_dsp_clear(&s->vd);
+ vorbis_info_clear(&s->vi);
+
+ av_fifo_free(s->pkt_fifo);
+ ff_af_queue_close(&s->afq);
+#if FF_API_OLD_ENCODE_AUDIO
+ av_freep(&avctx->coded_frame);
+#endif
+ av_freep(&avctx->extradata);
+
+ return 0;
}
-static int oggvorbis_encode_init(AVCodecContext *avccontext) {
- OggVorbisContext *context = avccontext->priv_data ;
+static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
+{
+ OggVorbisContext *s = avctx->priv_data;
ogg_packet header, header_comm, header_code;
uint8_t *p;
- unsigned int offset, len;
+ unsigned int offset;
+ int ret;
- 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_info_init(&s->vi);
+ if ((ret = oggvorbis_init_encoder(&s->vi, avctx))) {
+ av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n");
+ goto error;
+ }
+ if ((ret = vorbis_analysis_init(&s->vd, &s->vi))) {
+ av_log(avctx, AV_LOG_ERROR, "analysis init failed\n");
+ ret = vorbis_error_to_averror(ret);
+ goto error;
+ }
+ s->dsp_initialized = 1;
+ if ((ret = vorbis_block_init(&s->vd, &s->vb))) {
+ av_log(avctx, AV_LOG_ERROR, "dsp init failed\n");
+ ret = vorbis_error_to_averror(ret);
+ goto error;
}
- 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_init(&s->vc);
+ vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT);
- vorbis_analysis_headerout(&context->vd, &context->vc, &header,
- &header_comm, &header_code);
+ if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm,
+ &header_code))) {
+ ret = vorbis_error_to_averror(ret);
+ goto error;
+ }
- 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;
+ avctx->extradata_size = 1 + xiph_len(header.bytes) +
+ xiph_len(header_comm.bytes) +
+ header_code.bytes;
+ p = avctx->extradata = av_malloc(avctx->extradata_size +
+ FF_INPUT_BUFFER_PADDING_SIZE);
+ if (!p) {
+ ret = AVERROR(ENOMEM);
+ goto error;
+ }
+ 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_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);
+ assert(offset == avctx->extradata_size);
- 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;
- }
+ if ((ret = avpriv_vorbis_parse_extradata(avctx, &s->vp)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "invalid extradata\n");
+ return ret;
}
- 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);
- }
- }
+ vorbis_comment_clear(&s->vc);
- l=0;
- if(context->buffer_index){
- ogg_packet *op2= (ogg_packet*)context->buffer;
- op2->packet = context->buffer + sizeof(ogg_packet);
+ avctx->frame_size = OGGVORBIS_FRAME_SIZE;
+ ff_af_queue_init(avctx, &s->afq);
- 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);
+ s->pkt_fifo = av_fifo_alloc(BUFFER_SIZE);
+ if (!s->pkt_fifo) {
+ ret = AVERROR(ENOMEM);
+ goto error;
}
- 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);
+#if FF_API_OLD_ENCODE_AUDIO
+ avctx->coded_frame = avcodec_alloc_frame();
+ if (!avctx->coded_frame) {
+ ret = AVERROR(ENOMEM);
+ goto error;
+ }
+#endif
- return 0 ;
+ return 0;
+error:
+ oggvorbis_encode_close(avctx);
+ return ret;
}
-
-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];
+static int oggvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
+{
+ OggVorbisContext *s = avctx->priv_data;
+ ogg_packet op;
+ int ret, duration;
+
+ /* send samples to libvorbis */
+ if (frame) {
+ const float *audio = (const float *)frame->data[0];
+ const int samples = frame->nb_samples;
+ float **buffer;
+ int c, channels = s->vi.channels;
+
+ buffer = vorbis_analysis_buffer(&s->vd, samples);
+ for (c = 0; c < channels; c++) {
+ int i;
+ int co = (channels > 8) ? c :
+ ff_vorbis_encoding_channel_layout_offsets[channels - 1][c];
+ for (i = 0; i < samples; i++)
+ buffer[c][i] = audio[i * channels + co];
}
- } 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++;
+ if ((ret = vorbis_analysis_wrote(&s->vd, samples)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
+ return vorbis_error_to_averror(ret);
}
- 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];
+ if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
+ return ret;
} else {
- av_log(avccontext, AV_LOG_ERROR,
- "vorbis initial header len is wrong: %d\n", *p);
- return -1;
+ if (!s->eof)
+ if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
+ return vorbis_error_to_averror(ret);
+ }
+ s->eof = 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;
+ /* retrieve available packets from libvorbis */
+ while ((ret = vorbis_analysis_blockout(&s->vd, &s->vb)) == 1) {
+ if ((ret = vorbis_analysis(&s->vb, NULL)) < 0)
+ break;
+ if ((ret = vorbis_bitrate_addblock(&s->vb)) < 0)
+ break;
+
+ /* add any available packets to the output packet buffer */
+ while ((ret = vorbis_bitrate_flushpacket(&s->vd, &op)) == 1) {
+ if (av_fifo_space(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) {
+ av_log(avctx, AV_LOG_ERROR, "packet buffer is too small");
+ return AVERROR_BUG;
+ }
+ av_fifo_generic_write(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
+ av_fifo_generic_write(s->pkt_fifo, op.packet, op.bytes, NULL);
}
- }
-
- 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;
-
- val = av_clip_int16(val);
-
- *ptr = val ;
- ptr += channels;
+ if (ret < 0) {
+ av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
+ break;
}
}
-
- 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;
+ if (ret < 0) {
+ av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
+ return vorbis_error_to_averror(ret);
}
- 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) ;
+ /* check for available packets */
+ if (av_fifo_size(s->pkt_fifo) < sizeof(ogg_packet))
+ return 0;
- total_samples = 0 ;
- total_bytes = 0 ;
+ av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
- 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) ;
+ if ((ret = ff_alloc_packet(avpkt, op.bytes))) {
+ av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+ return ret;
+ }
+ av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL);
+
+ avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos);
+
+ duration = avpriv_vorbis_parse_frame(&s->vp, avpkt->data, avpkt->size);
+ if (duration > 0) {
+ /* we do not know encoder delay until we get the first packet from
+ * libvorbis, so we have to update the AudioFrameQueue counts */
+ if (!avctx->delay) {
+ avctx->delay = duration;
+ s->afq.remaining_delay += duration;
+ s->afq.remaining_samples += duration;
+ }
+ ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);
}
- *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 ;
+ *got_packet_ptr = 1;
+ 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,
-} ;
+AVCodec ff_libvorbis_encoder = {
+ .name = "libvorbis",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_VORBIS,
+ .priv_data_size = sizeof(OggVorbisContext),
+ .init = oggvorbis_encode_init,
+ .encode2 = oggvorbis_encode_frame,
+ .close = oggvorbis_encode_close,
+ .capabilities = CODEC_CAP_DELAY,
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
+ AV_SAMPLE_FMT_NONE },
+ .long_name = NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),
+ .priv_class = &class,
+ .defaults = defaults,
+};
projects / ffmpeg / blobdiff