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ffmpeg/libavcodec/libvorbisenc.c

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/*
* Copyright (c) 2002 Mark Hills <mark@pogo.org.uk>
*
* This file is part of FFmpeg.
*
* FFmpeg 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,
* 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
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <vorbis/vorbisenc.h>
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/fifo.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "encode.h"
#include "internal.h"
#include "vorbis.h"
#include "vorbis_parser.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 LIBVORBIS_FRAME_SIZE 64
#define BUFFER_SIZE (1024 * 64)
typedef struct LibvorbisEncContext {
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 */
AVFifo *pkt_fifo; /**< output packet buffer */
int eof; /**< end-of-file flag */
int dsp_initialized; /**< vd has been initialized */
vorbis_comment vc; /**< VorbisComment info */
double iblock; /**< impulse block bias option */
AVVorbisParseContext *vp; /**< parse context to get durations */
AudioFrameQueue afq; /**< frame queue for timestamps */
} LibvorbisEncContext;
static const AVOption options[] = {
{ "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 },
{ NULL }
};
static const AVCodecDefault defaults[] = {
{ "b", "0" },
{ NULL },
};
static const AVClass vorbis_class = {
.class_name = "libvorbis",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
static const uint8_t vorbis_encoding_channel_layout_offsets[8][8] = {
{ 0 },
{ 0, 1 },
{ 0, 2, 1 },
{ 0, 1, 2, 3 },
{ 0, 2, 1, 3, 4 },
{ 0, 2, 1, 4, 5, 3 },
{ 0, 2, 1, 5, 6, 4, 3 },
{ 0, 2, 1, 6, 7, 4, 5, 3 },
};
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 av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx)
{
LibvorbisEncContext *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; /* should not happen */
}
/* cutoff frequency */
if (avctx->cutoff > 0) {
cfreq = avctx->cutoff / 1000.0;
if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))
goto error; /* should not happen */
}
/* impulse block bias */
if (s->iblock) {
if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock)))
goto error;
}
if (avctx->channels == 3 &&
avctx->channel_layout != (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER) ||
avctx->channels == 4 &&
avctx->channel_layout != AV_CH_LAYOUT_2_2 &&
avctx->channel_layout != AV_CH_LAYOUT_QUAD ||
avctx->channels == 5 &&
avctx->channel_layout != AV_CH_LAYOUT_5POINT0 &&
avctx->channel_layout != AV_CH_LAYOUT_5POINT0_BACK ||
avctx->channels == 6 &&
avctx->channel_layout != AV_CH_LAYOUT_5POINT1 &&
avctx->channel_layout != AV_CH_LAYOUT_5POINT1_BACK ||
avctx->channels == 7 &&
avctx->channel_layout != (AV_CH_LAYOUT_5POINT1|AV_CH_BACK_CENTER) ||
avctx->channels == 8 &&
avctx->channel_layout != AV_CH_LAYOUT_7POINT1) {
if (avctx->channel_layout) {
char name[32];
av_get_channel_layout_string(name, sizeof(name), avctx->channels,
avctx->channel_layout);
av_log(avctx, AV_LOG_ERROR, "%s not supported by Vorbis: "
"output stream will have incorrect "
"channel layout.\n", name);
} else {
av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The encoder "
"will use Vorbis channel layout for "
"%d channels.\n", avctx->channels);
}
}
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 libvorbis_encode_close(AVCodecContext *avctx)
{
LibvorbisEncContext *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_freep2(&s->pkt_fifo);
ff_af_queue_close(&s->afq);
av_vorbis_parse_free(&s->vp);
return 0;
}
static av_cold int libvorbis_encode_init(AVCodecContext *avctx)
{
LibvorbisEncContext *s = avctx->priv_data;
ogg_packet header, header_comm, header_code;
uint8_t *p;
unsigned int offset;
int ret;
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_comment_init(&s->vc);
if (!(avctx->flags & AV_CODEC_FLAG_BITEXACT))
vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT);
if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm,
&header_code))) {
ret = vorbis_error_to_averror(ret);
goto error;
}
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.bytes;
memcpy(&p[offset], header_comm.packet, header_comm.bytes);
offset += header_comm.bytes;
memcpy(&p[offset], header_code.packet, header_code.bytes);
offset += header_code.bytes;
av_assert0(offset == avctx->extradata_size);
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;
}
vorbis_comment_clear(&s->vc);
avctx->frame_size = LIBVORBIS_FRAME_SIZE;
ff_af_queue_init(avctx, &s->afq);
s->pkt_fifo = av_fifo_alloc2(BUFFER_SIZE, 1, 0);
if (!s->pkt_fifo) {
ret = AVERROR(ENOMEM);
goto error;
}
return 0;
error:
libvorbis_encode_close(avctx);
return ret;
}
static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
LibvorbisEncContext *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 :
vorbis_encoding_channel_layout_offsets[channels - 1][c];
memcpy(buffer[c], frame->extended_data[co],
samples * sizeof(*buffer[c]));
}
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 && s->afq.frame_alloc)
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;
}
/* 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_can_write(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) {
av_log(avctx, AV_LOG_ERROR, "packet buffer is too small\n");
return AVERROR_BUG;
}
av_fifo_write(s->pkt_fifo, &op, sizeof(ogg_packet));
av_fifo_write(s->pkt_fifo, op.packet, op.bytes);
}
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);
}
/* Read an available packet if possible */
if (av_fifo_read(s->pkt_fifo, &op, sizeof(ogg_packet)) < 0)
return 0;
if ((ret = ff_get_encode_buffer(avctx, avpkt, op.bytes, 0)) < 0)
return ret;
av_fifo_read(s->pkt_fifo, avpkt->data, op.bytes);
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;
}
const AVCodec ff_libvorbis_encoder = {
.name = "libvorbis",
.long_name = NULL_IF_CONFIG_SMALL("libvorbis"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_VORBIS,
.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
AV_CODEC_CAP_SMALL_LAST_FRAME,
.priv_data_size = sizeof(LibvorbisEncContext),
.init = libvorbis_encode_init,
.encode2 = libvorbis_encode_frame,
.close = libvorbis_encode_close,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.priv_class = &vorbis_class,
.defaults = defaults,
.wrapper_name = "libvorbis",
};
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