/*
* 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 libavcodec/libvorbis.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>
-#define OGGVORBIS_FRAME_SIZE 64
+/* 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 LIBVORBIS_FRAME_SIZE 64
+
+#define BUFFER_SIZE (1024 * 64)
+
+typedef struct LibvorbisContext {
+ 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 */
+ AVVorbisParseContext *vp; /**< parse context to get durations */
+ AudioFrameQueue afq; /**< frame queue for timestamps */
+} LibvorbisContext;
+
+static const AVOption options[] = {
+ { "iblock", "Sets the impulse block bias", offsetof(LibvorbisContext, 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;
+ }
+}
-#define BUFFER_SIZE (1024*64)
+static av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx)
+{
+ LibvorbisContext *s = avctx->priv_data;
+ double cfreq;
+ int ret;
+
+ if (avctx->flags & AV_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 & AV_CODEC_FLAG_QSCALE))
+ q = 3.0;
+ if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels,
+ avctx->sample_rate,
+ q / 10.0)))
+ goto error;
+ } else {
+ 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;
+ }
-typedef struct OggVorbisContext {
- vorbis_info vi ;
- vorbis_dsp_state vd ;
- vorbis_block vb ;
- uint8_t buffer[BUFFER_SIZE];
- int buffer_index;
- int eof;
+ /* cutoff frequency */
+ if (avctx->cutoff > 0) {
+ cfreq = avctx->cutoff / 1000.0;
+ if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))
+ goto error;
+ }
- /* decoder */
- vorbis_comment vc ;
- ogg_packet op;
-} OggVorbisContext ;
+ /* 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;
-static int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avccontext) {
- double cfreq;
+ return 0;
+error:
+ return vorbis_error_to_averror(ret);
+}
- 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))
- return -1;
- } else {
- /* constant bitrate */
- if(vorbis_encode_setup_managed(vi, avccontext->channels,
- avccontext->sample_rate, -1, avccontext->bit_rate, -1))
- return -1;
+/* How many bytes are needed for a buffer of length 'l' */
+static int xiph_len(int l)
+{
+ return 1 + l / 255 + l;
+}
-#ifdef OGGVORBIS_VBR_BY_ESTIMATE
- /* variable bitrate by estimate */
- if(vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE_AVG, NULL))
- return -1;
-#endif
- }
+static av_cold int libvorbis_encode_close(AVCodecContext *avctx)
+{
+ LibvorbisContext *s = avctx->priv_data;
- /* cutoff frequency */
- if(avccontext->cutoff > 0) {
- cfreq = avccontext->cutoff / 1000.0;
- if(vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq))
- return -1;
- }
+ /* 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);
+ av_freep(&avctx->extradata);
- return vorbis_encode_setup_init(vi);
+ av_vorbis_parse_free(&s->vp);
+
+ return 0;
}
-static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext) {
- OggVorbisContext *context = avccontext->priv_data ;
+static av_cold int libvorbis_encode_init(AVCodecContext *avctx)
+{
+ LibvorbisContext *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 = libvorbis_setup(&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 +
+ AV_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);
+ assert(offset == avctx->extradata_size);
-/* vorbis_block_clear(&context->vb);
- vorbis_dsp_clear(&context->vd);
- vorbis_info_clear(&context->vi);*/
- vorbis_comment_clear(&context->vc);
+ s->vp = av_vorbis_parse_init(avctx->extradata, avctx->extradata_size);
+ if (!s->vp) {
+ av_log(avctx, AV_LOG_ERROR, "invalid extradata\n");
+ return ret;
+ }
- avccontext->frame_size = OGGVORBIS_FRAME_SIZE ;
+ vorbis_comment_clear(&s->vc);
- avccontext->coded_frame= avcodec_alloc_frame();
- avccontext->coded_frame->key_frame= 1;
+ avctx->frame_size = LIBVORBIS_FRAME_SIZE;
+ ff_af_queue_init(avctx, &s->afq);
- return 0 ;
-}
+ s->pkt_fifo = av_fifo_alloc(BUFFER_SIZE);
+ if (!s->pkt_fifo) {
+ ret = AVERROR(ENOMEM);
+ goto error;
+ }
+ return 0;
+error:
+ libvorbis_encode_close(avctx);
+ return ret;
+}
-static int oggvorbis_encode_frame(AVCodecContext *avccontext,
- unsigned char *packets,
- int buf_size, void *data)
+static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
{
- 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;
- }
+ LibvorbisContext *s = avctx->priv_data;
+ ogg_packet op;
+ int ret, duration;
+
+ /* send samples to libvorbis */
+ if (frame) {
+ 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 co = (channels > 8) ? c :
+ ff_vorbis_encoding_channel_layout_offsets[channels - 1][c];
+ memcpy(buffer[c], frame->extended_data[co],
+ samples * sizeof(*buffer[c]));
}
- 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)
- 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);
+ 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);
}
+ if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
+ return ret;
+ } else {
+ 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;
}
- 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
-
- 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);
+ /* 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);
+ }
+ if (ret < 0) {
+ av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
+ break;
+ }
+ }
+ if (ret < 0) {
+ av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
+ return vorbis_error_to_averror(ret);
}
- 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 */
+ /* check for available packets */
+ if (av_fifo_size(s->pkt_fifo) < sizeof(ogg_packet))
+ return 0;
- vorbis_block_clear(&context->vb);
- vorbis_dsp_clear(&context->vd);
- vorbis_info_clear(&context->vi);
+ av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
- av_freep(&avccontext->coded_frame);
- av_freep(&avccontext->extradata);
+ 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 = av_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->initial_padding) {
+ avctx->initial_padding = duration;
+ s->afq.remaining_delay += duration;
+ s->afq.remaining_samples += duration;
+ }
+ ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);
+ }
- return 0 ;
+ *got_packet_ptr = 1;
+ return 0;
}
-
-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,
- .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
- .long_name= NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),
-} ;
+AVCodec ff_libvorbis_encoder = {
+ .name = "libvorbis",
+ .long_name = NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_VORBIS,
+ .priv_data_size = sizeof(LibvorbisContext),
+ .init = libvorbis_encode_init,
+ .encode2 = libvorbis_encode_frame,
+ .close = libvorbis_encode_close,
+ .capabilities = AV_CODEC_CAP_DELAY,
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
+ .priv_class = &class,
+ .defaults = defaults,
+};
projects / ffmpeg / blobdiff