mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-23 12:43:46 +02:00
47687a2f8a
Explicitly identify decoder/encoder wrappers with a common name. This saves API users from guessing by the name suffix. For example, they don't have to guess that "h264_qsv" is the h264 QSV implementation, and instead they can just check the AVCodec .codec and .wrapper_name fields. Explicitly mark AVCodec entries that are hardware decoders or most likely hardware decoders with new AV_CODEC_CAPs. The purpose is allowing API users listing hardware decoders in a more generic way. The proposed AVCodecHWConfig does not provide this information fully, because it's concerned with decoder configuration, not information about the fact whether the hardware is used or not. AV_CODEC_CAP_HYBRID exists specifically for QSV, which can have software implementations in case the hardware is not capable. Based on a patch by Philip Langdale <philipl@overt.org>. Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
361 lines
12 KiB
C
361 lines
12 KiB
C
/*
|
|
* copyright (c) 2002 Mark Hills <mark@pogo.org.uk>
|
|
*
|
|
* This file is part of Libav.
|
|
*
|
|
* 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.
|
|
*
|
|
* 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 Libav; if not, write to the Free Software
|
|
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
|
*/
|
|
|
|
/**
|
|
* @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>
|
|
|
|
/* 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 = {
|
|
.class_name = "libvorbis",
|
|
.item_name = av_default_item_name,
|
|
.option = options,
|
|
.version = 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;
|
|
}
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
/* cutoff frequency */
|
|
if (avctx->cutoff > 0) {
|
|
cfreq = avctx->cutoff / 1000.0;
|
|
if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))
|
|
goto error;
|
|
}
|
|
|
|
/* 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;
|
|
|
|
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)
|
|
{
|
|
LibvorbisContext *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_free(s->pkt_fifo);
|
|
ff_af_queue_close(&s->afq);
|
|
av_freep(&avctx->extradata);
|
|
|
|
av_vorbis_parse_free(&s->vp);
|
|
|
|
return 0;
|
|
}
|
|
|
|
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;
|
|
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);
|
|
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;
|
|
assert(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_alloc(BUFFER_SIZE);
|
|
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)
|
|
{
|
|
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]));
|
|
}
|
|
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;
|
|
}
|
|
|
|
/* 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);
|
|
}
|
|
|
|
/* check for available packets */
|
|
if (av_fifo_size(s->pkt_fifo) < sizeof(ogg_packet))
|
|
return 0;
|
|
|
|
av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
|
|
|
|
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);
|
|
}
|
|
|
|
*got_packet_ptr = 1;
|
|
return 0;
|
|
}
|
|
|
|
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,
|
|
.wrapper_name = "libvorbis",
|
|
};
|