mirror of
https://git.videolan.org/git/ffmpeg.git
synced 2024-08-22 09:15:06 +02:00
562 lines
19 KiB
C
562 lines
19 KiB
C
/*
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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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/**
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* @file
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* Vorbis encoding support via libvorbisenc.
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* @author Mark Hills <mark@pogo.org.uk>
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*/
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#include <vorbis/vorbisenc.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 "bytestream.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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#undef NDEBUG
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#include <assert.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 OGGVORBIS_FRAME_SIZE 64
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#define BUFFER_SIZE (1024 * 64)
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typedef struct OggVorbisContext {
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AVClass *av_class; /**< class for AVOptions */
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AVFrame frame;
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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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ogg_packet op; /**< ogg packet */
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double iblock; /**< impulse block bias option */
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VorbisParseContext vp; /**< parse context to get durations */
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AudioFrameQueue afq; /**< frame queue for timestamps */
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} OggVorbisContext;
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static const AVOption options[] = {
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{ "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 },
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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 class = { "libvorbis", av_default_item_name, options, LIBAVUTIL_VERSION_INT };
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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 oggvorbis_init_encoder(vorbis_info *vi,
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AVCodecContext *avctx)
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{
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OggVorbisContext *s = avctx->priv_data;
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double cfreq;
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int ret;
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if (avctx->flags & 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 & 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 oggvorbis_encode_close(AVCodecContext *avctx)
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{
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OggVorbisContext *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_free(s->pkt_fifo);
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ff_af_queue_close(&s->afq);
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#if FF_API_OLD_ENCODE_AUDIO
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av_freep(&avctx->coded_frame);
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#endif
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av_freep(&avctx->extradata);
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return 0;
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}
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static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
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{
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OggVorbisContext *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 = oggvorbis_init_encoder(&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 & 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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FF_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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assert(offset == avctx->extradata_size);
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if ((ret = avpriv_vorbis_parse_extradata(avctx, &s->vp)) < 0) {
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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 = OGGVORBIS_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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#if FF_API_OLD_ENCODE_AUDIO
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avctx->coded_frame = avcodec_alloc_frame();
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if (!avctx->coded_frame) {
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ret = AVERROR(ENOMEM);
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goto error;
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}
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#endif
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return 0;
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error:
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oggvorbis_encode_close(avctx);
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return ret;
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}
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static int oggvorbis_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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OggVorbisContext *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 float *audio = (const float *)frame->data[0];
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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 i;
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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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for (i = 0; i < samples; i++)
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buffer[c][i] = audio[i * channels + co];
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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)
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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)))
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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);
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duration = avpriv_vorbis_parse_frame(&s->vp, avpkt->data, avpkt->size);
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if (duration > 0) {
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/* we do not know encoder delay until we get the first packet from
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* libvorbis, so we have to update the AudioFrameQueue counts */
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if (!avctx->delay) {
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avctx->delay = duration;
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s->afq.remaining_delay += duration;
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s->afq.remaining_samples += duration;
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}
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ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);
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}
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*got_packet_ptr = 1;
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return 0;
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}
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AVCodec ff_libvorbis_encoder = {
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.name = "libvorbis",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_VORBIS,
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.priv_data_size = sizeof(OggVorbisContext),
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.init = oggvorbis_encode_init,
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.encode2 = oggvorbis_encode_frame,
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.close = oggvorbis_encode_close,
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.capabilities = CODEC_CAP_DELAY,
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_NONE },
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.long_name = NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),
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.priv_class = &class,
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.defaults = defaults,
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};
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static int oggvorbis_decode_init(AVCodecContext *avccontext) {
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OggVorbisContext *context = avccontext->priv_data ;
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uint8_t *p= avccontext->extradata;
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int i, hsizes[3];
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unsigned char *headers[3], *extradata = avccontext->extradata;
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vorbis_info_init(&context->vi) ;
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vorbis_comment_init(&context->vc) ;
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if(! avccontext->extradata_size || ! p) {
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av_log(avccontext, AV_LOG_ERROR, "vorbis extradata absent\n");
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return -1;
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}
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if(p[0] == 0 && p[1] == 30) {
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for(i = 0; i < 3; i++){
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hsizes[i] = bytestream_get_be16(&p);
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headers[i] = p;
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p += hsizes[i];
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}
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} else if(*p == 2) {
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unsigned int offset = 1;
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p++;
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for(i=0; i<2; i++) {
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hsizes[i] = 0;
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while((*p == 0xFF) && (offset < avccontext->extradata_size)) {
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hsizes[i] += 0xFF;
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offset++;
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p++;
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}
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if(offset >= avccontext->extradata_size - 1) {
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av_log(avccontext, AV_LOG_ERROR,
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"vorbis header sizes damaged\n");
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return -1;
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}
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hsizes[i] += *p;
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offset++;
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p++;
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}
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hsizes[2] = avccontext->extradata_size - hsizes[0]-hsizes[1]-offset;
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#if 0
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av_log(avccontext, AV_LOG_DEBUG,
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"vorbis header sizes: %d, %d, %d, / extradata_len is %d \n",
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hsizes[0], hsizes[1], hsizes[2], avccontext->extradata_size);
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#endif
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headers[0] = extradata + offset;
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headers[1] = extradata + offset + hsizes[0];
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headers[2] = extradata + offset + hsizes[0] + hsizes[1];
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} else {
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av_log(avccontext, AV_LOG_ERROR,
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"vorbis initial header len is wrong: %d\n", *p);
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return -1;
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}
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for(i=0; i<3; i++){
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context->op.b_o_s= i==0;
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context->op.bytes = hsizes[i];
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context->op.packet = headers[i];
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if(vorbis_synthesis_headerin(&context->vi, &context->vc, &context->op)<0){
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av_log(avccontext, AV_LOG_ERROR, "%d. vorbis header damaged\n", i+1);
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return -1;
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}
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}
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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;
|
|
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++) {
|
|
*ptr = av_clip_int16(mono[j] * 32767.f);
|
|
ptr += channels;
|
|
}
|
|
}
|
|
|
|
return 0 ;
|
|
}
|
|
|
|
static int oggvorbis_decode_frame(AVCodecContext *avccontext, void *data,
|
|
int *got_frame_ptr, AVPacket *avpkt)
|
|
{
|
|
OggVorbisContext *context = avccontext->priv_data ;
|
|
float **pcm ;
|
|
ogg_packet *op= &context->op;
|
|
int samples, total_samples, total_bytes;
|
|
int ret;
|
|
int16_t *output;
|
|
|
|
if(!avpkt->size){
|
|
//FIXME flush
|
|
return 0;
|
|
}
|
|
|
|
context->frame.nb_samples = 8192*4;
|
|
if ((ret = avccontext->get_buffer(avccontext, &context->frame)) < 0) {
|
|
av_log(avccontext, AV_LOG_ERROR, "get_buffer() failed\n");
|
|
return ret;
|
|
}
|
|
output = (int16_t *)context->frame.data[0];
|
|
|
|
|
|
op->packet = avpkt->data;
|
|
op->bytes = avpkt->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) ;
|
|
|
|
total_samples = 0 ;
|
|
total_bytes = 0 ;
|
|
|
|
while((samples = vorbis_synthesis_pcmout(&context->vd, &pcm)) > 0) {
|
|
conv(samples, pcm, (char*)output + total_bytes, context->vi.channels) ;
|
|
total_bytes += samples * 2 * context->vi.channels ;
|
|
total_samples += samples ;
|
|
vorbis_synthesis_read(&context->vd, samples) ;
|
|
}
|
|
|
|
context->frame.nb_samples = total_samples;
|
|
*got_frame_ptr = 1;
|
|
*(AVFrame *)data = context->frame;
|
|
return avpkt->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 ;
|
|
}
|
|
|
|
|
|
AVCodec ff_libvorbis_decoder = {
|
|
.name = "libvorbis",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = CODEC_ID_VORBIS,
|
|
.priv_data_size = sizeof(OggVorbisContext),
|
|
.init = oggvorbis_decode_init,
|
|
.decode = oggvorbis_decode_frame,
|
|
.close = oggvorbis_decode_close,
|
|
.capabilities = CODEC_CAP_DELAY,
|
|
} ;
|
|
|