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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00
FFmpeg/libavcodec/libvorbisenc.c
Andreas Rheinhardt 790f793844 avutil/common: Don't auto-include mem.h
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.

Keep it for external users in order to not cause breakages.

Also improve the other headers a bit while just at it.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-31 00:08:43 +01:00

395 lines
14 KiB
C

/*
* 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/mem.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "codec_internal.h"
#include "encode.h"
#include "version.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 FFCodecDefault 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;
int channels = avctx->ch_layout.nb_channels;
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, 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, 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 ((channels == 3 &&
av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_SURROUND)) ||
(channels == 4 &&
av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_2_2) &&
av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_QUAD)) ||
(channels == 5 &&
av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT0) &&
av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT0_BACK)) ||
(channels == 6 &&
av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT1) &&
av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT1_BACK)) ||
(channels == 7 &&
av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_6POINT1)) ||
(channels == 8 &&
av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_7POINT1))) {
if (avctx->ch_layout.order != AV_CHANNEL_ORDER_UNSPEC) {
char name[32];
av_channel_layout_describe(&avctx->ch_layout, name, sizeof(name));
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", 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 FFCodec ff_libvorbis_encoder = {
.p.name = "libvorbis",
CODEC_LONG_NAME("libvorbis"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_VORBIS,
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
AV_CODEC_CAP_SMALL_LAST_FRAME,
.caps_internal = FF_CODEC_CAP_NOT_INIT_THREADSAFE,
.priv_data_size = sizeof(LibvorbisEncContext),
.init = libvorbis_encode_init,
FF_CODEC_ENCODE_CB(libvorbis_encode_frame),
.close = libvorbis_encode_close,
.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.p.priv_class = &vorbis_class,
.defaults = defaults,
.p.wrapper_name = "libvorbis",
};