forked from KolibriOS/kolibrios
381 lines
14 KiB
C
381 lines
14 KiB
C
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/*
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* Copyright (c) 2002 Mark Hills <mark@pogo.org.uk>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <vorbis/vorbisenc.h>
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#include "libavutil/avassert.h"
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#include "libavutil/fifo.h"
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "audio_frame_queue.h"
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#include "internal.h"
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#include "vorbis.h"
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#include "vorbis_parser.h"
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/* Number of samples the user should send in each call.
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* This value is used because it is the LCD of all possible frame sizes, so
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* an output packet will always start at the same point as one of the input
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* packets.
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*/
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#define LIBVORBIS_FRAME_SIZE 64
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#define BUFFER_SIZE (1024 * 64)
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typedef struct LibvorbisEncContext {
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AVClass *av_class; /**< class for AVOptions */
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vorbis_info vi; /**< vorbis_info used during init */
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vorbis_dsp_state vd; /**< DSP state used for analysis */
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vorbis_block vb; /**< vorbis_block used for analysis */
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AVFifoBuffer *pkt_fifo; /**< output packet buffer */
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int eof; /**< end-of-file flag */
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int dsp_initialized; /**< vd has been initialized */
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vorbis_comment vc; /**< VorbisComment info */
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double iblock; /**< impulse block bias option */
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AVVorbisParseContext *vp; /**< parse context to get durations */
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AudioFrameQueue afq; /**< frame queue for timestamps */
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} LibvorbisEncContext;
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static const AVOption options[] = {
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{ "iblock", "Sets the impulse block bias", offsetof(LibvorbisEncContext, iblock), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -15, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
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{ NULL }
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};
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static const AVCodecDefault defaults[] = {
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{ "b", "0" },
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{ NULL },
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};
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static const AVClass vorbis_class = {
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.class_name = "libvorbis",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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static int vorbis_error_to_averror(int ov_err)
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{
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switch (ov_err) {
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case OV_EFAULT: return AVERROR_BUG;
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case OV_EINVAL: return AVERROR(EINVAL);
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case OV_EIMPL: return AVERROR(EINVAL);
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default: return AVERROR_UNKNOWN;
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}
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}
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static av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx)
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{
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LibvorbisEncContext *s = avctx->priv_data;
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double cfreq;
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int ret;
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if (avctx->flags & AV_CODEC_FLAG_QSCALE || !avctx->bit_rate) {
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/* variable bitrate
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* NOTE: we use the oggenc range of -1 to 10 for global_quality for
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* user convenience, but libvorbis uses -0.1 to 1.0.
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*/
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float q = avctx->global_quality / (float)FF_QP2LAMBDA;
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/* default to 3 if the user did not set quality or bitrate */
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if (!(avctx->flags & AV_CODEC_FLAG_QSCALE))
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q = 3.0;
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if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels,
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avctx->sample_rate,
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q / 10.0)))
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goto error;
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} else {
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int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1;
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int maxrate = avctx->rc_max_rate > 0 ? avctx->rc_max_rate : -1;
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/* average bitrate */
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if ((ret = vorbis_encode_setup_managed(vi, avctx->channels,
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avctx->sample_rate, maxrate,
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avctx->bit_rate, minrate)))
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goto error;
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/* variable bitrate by estimate, disable slow rate management */
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if (minrate == -1 && maxrate == -1)
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if ((ret = vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL)))
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goto error; /* should not happen */
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}
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/* cutoff frequency */
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if (avctx->cutoff > 0) {
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cfreq = avctx->cutoff / 1000.0;
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if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))
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goto error; /* should not happen */
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}
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/* impulse block bias */
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if (s->iblock) {
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if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock)))
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goto error;
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}
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if (avctx->channels == 3 &&
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avctx->channel_layout != (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER) ||
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avctx->channels == 4 &&
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avctx->channel_layout != AV_CH_LAYOUT_2_2 &&
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avctx->channel_layout != AV_CH_LAYOUT_QUAD ||
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avctx->channels == 5 &&
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avctx->channel_layout != AV_CH_LAYOUT_5POINT0 &&
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avctx->channel_layout != AV_CH_LAYOUT_5POINT0_BACK ||
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avctx->channels == 6 &&
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avctx->channel_layout != AV_CH_LAYOUT_5POINT1 &&
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avctx->channel_layout != AV_CH_LAYOUT_5POINT1_BACK ||
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avctx->channels == 7 &&
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avctx->channel_layout != (AV_CH_LAYOUT_5POINT1|AV_CH_BACK_CENTER) ||
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avctx->channels == 8 &&
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avctx->channel_layout != AV_CH_LAYOUT_7POINT1) {
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if (avctx->channel_layout) {
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char name[32];
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av_get_channel_layout_string(name, sizeof(name), avctx->channels,
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avctx->channel_layout);
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av_log(avctx, AV_LOG_ERROR, "%s not supported by Vorbis: "
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"output stream will have incorrect "
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"channel layout.\n", name);
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} else {
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av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The encoder "
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"will use Vorbis channel layout for "
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"%d channels.\n", avctx->channels);
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}
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}
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if ((ret = vorbis_encode_setup_init(vi)))
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goto error;
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return 0;
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error:
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return vorbis_error_to_averror(ret);
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}
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/* How many bytes are needed for a buffer of length 'l' */
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static int xiph_len(int l)
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{
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return 1 + l / 255 + l;
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}
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static av_cold int libvorbis_encode_close(AVCodecContext *avctx)
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{
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LibvorbisEncContext *s = avctx->priv_data;
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/* notify vorbisenc this is EOF */
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if (s->dsp_initialized)
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vorbis_analysis_wrote(&s->vd, 0);
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vorbis_block_clear(&s->vb);
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vorbis_dsp_clear(&s->vd);
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vorbis_info_clear(&s->vi);
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av_fifo_freep(&s->pkt_fifo);
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ff_af_queue_close(&s->afq);
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av_freep(&avctx->extradata);
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av_vorbis_parse_free(&s->vp);
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return 0;
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}
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static av_cold int libvorbis_encode_init(AVCodecContext *avctx)
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{
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LibvorbisEncContext *s = avctx->priv_data;
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ogg_packet header, header_comm, header_code;
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uint8_t *p;
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unsigned int offset;
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int ret;
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vorbis_info_init(&s->vi);
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if ((ret = libvorbis_setup(&s->vi, avctx))) {
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av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n");
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goto error;
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}
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if ((ret = vorbis_analysis_init(&s->vd, &s->vi))) {
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av_log(avctx, AV_LOG_ERROR, "analysis init failed\n");
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ret = vorbis_error_to_averror(ret);
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goto error;
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}
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s->dsp_initialized = 1;
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if ((ret = vorbis_block_init(&s->vd, &s->vb))) {
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av_log(avctx, AV_LOG_ERROR, "dsp init failed\n");
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ret = vorbis_error_to_averror(ret);
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goto error;
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}
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vorbis_comment_init(&s->vc);
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if (!(avctx->flags & AV_CODEC_FLAG_BITEXACT))
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vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT);
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if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm,
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&header_code))) {
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ret = vorbis_error_to_averror(ret);
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goto error;
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}
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avctx->extradata_size = 1 + xiph_len(header.bytes) +
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xiph_len(header_comm.bytes) +
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header_code.bytes;
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p = avctx->extradata = av_malloc(avctx->extradata_size +
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AV_INPUT_BUFFER_PADDING_SIZE);
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if (!p) {
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ret = AVERROR(ENOMEM);
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goto error;
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}
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p[0] = 2;
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offset = 1;
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offset += av_xiphlacing(&p[offset], header.bytes);
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offset += av_xiphlacing(&p[offset], header_comm.bytes);
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memcpy(&p[offset], header.packet, header.bytes);
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offset += header.bytes;
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memcpy(&p[offset], header_comm.packet, header_comm.bytes);
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offset += header_comm.bytes;
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memcpy(&p[offset], header_code.packet, header_code.bytes);
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offset += header_code.bytes;
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av_assert0(offset == avctx->extradata_size);
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s->vp = av_vorbis_parse_init(avctx->extradata, avctx->extradata_size);
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if (!s->vp) {
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av_log(avctx, AV_LOG_ERROR, "invalid extradata\n");
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return ret;
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}
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vorbis_comment_clear(&s->vc);
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avctx->frame_size = LIBVORBIS_FRAME_SIZE;
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ff_af_queue_init(avctx, &s->afq);
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s->pkt_fifo = av_fifo_alloc(BUFFER_SIZE);
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if (!s->pkt_fifo) {
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ret = AVERROR(ENOMEM);
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goto error;
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}
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return 0;
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error:
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libvorbis_encode_close(avctx);
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return ret;
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}
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static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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const AVFrame *frame, int *got_packet_ptr)
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{
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LibvorbisEncContext *s = avctx->priv_data;
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ogg_packet op;
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int ret, duration;
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/* send samples to libvorbis */
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if (frame) {
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const int samples = frame->nb_samples;
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float **buffer;
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int c, channels = s->vi.channels;
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buffer = vorbis_analysis_buffer(&s->vd, samples);
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for (c = 0; c < channels; c++) {
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int co = (channels > 8) ? c :
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ff_vorbis_encoding_channel_layout_offsets[channels - 1][c];
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memcpy(buffer[c], frame->extended_data[co],
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samples * sizeof(*buffer[c]));
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}
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if ((ret = vorbis_analysis_wrote(&s->vd, samples)) < 0) {
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av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
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return vorbis_error_to_averror(ret);
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}
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if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
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return ret;
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} else {
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if (!s->eof && s->afq.frame_alloc)
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if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) {
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av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
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return vorbis_error_to_averror(ret);
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}
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s->eof = 1;
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}
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/* retrieve available packets from libvorbis */
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while ((ret = vorbis_analysis_blockout(&s->vd, &s->vb)) == 1) {
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if ((ret = vorbis_analysis(&s->vb, NULL)) < 0)
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break;
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if ((ret = vorbis_bitrate_addblock(&s->vb)) < 0)
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break;
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/* add any available packets to the output packet buffer */
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while ((ret = vorbis_bitrate_flushpacket(&s->vd, &op)) == 1) {
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if (av_fifo_space(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) {
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av_log(avctx, AV_LOG_ERROR, "packet buffer is too small\n");
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return AVERROR_BUG;
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}
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av_fifo_generic_write(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
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av_fifo_generic_write(s->pkt_fifo, op.packet, op.bytes, NULL);
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}
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if (ret < 0) {
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av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
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break;
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}
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}
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if (ret < 0) {
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av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
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return vorbis_error_to_averror(ret);
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}
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/* check for available packets */
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if (av_fifo_size(s->pkt_fifo) < sizeof(ogg_packet))
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return 0;
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av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
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if ((ret = ff_alloc_packet2(avctx, avpkt, op.bytes, 0)) < 0)
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return ret;
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av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL);
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||
|
|
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 && s->afq.frames) {
|
||
|
|
avctx->initial_padding = duration;
|
||
|
|
av_assert0(!s->afq.remaining_delay);
|
||
|
|
s->afq.frames->duration += duration;
|
||
|
|
if (s->afq.frames->pts != AV_NOPTS_VALUE)
|
||
|
|
s->afq.frames->pts -= duration;
|
||
|
|
s->afq.remaining_samples += duration;
|
||
|
|
}
|
||
|
|
ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);
|
||
|
|
}
|
||
|
|
|
||
|
|
*got_packet_ptr = 1;
|
||
|
|
return 0;
|
||
|
|
}
|
||
|
|
|
||
|
|
AVCodec ff_libvorbis_encoder = {
|
||
|
|
.name = "libvorbis",
|
||
|
|
.long_name = NULL_IF_CONFIG_SMALL("libvorbis"),
|
||
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
||
|
|
.id = AV_CODEC_ID_VORBIS,
|
||
|
|
.priv_data_size = sizeof(LibvorbisEncContext),
|
||
|
|
.init = libvorbis_encode_init,
|
||
|
|
.encode2 = libvorbis_encode_frame,
|
||
|
|
.close = libvorbis_encode_close,
|
||
|
|
.capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_SMALL_LAST_FRAME,
|
||
|
|
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
|
||
|
|
AV_SAMPLE_FMT_NONE },
|
||
|
|
.priv_class = &vorbis_class,
|
||
|
|
.defaults = defaults,
|
||
|
|
};
|