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avcodec/pcm_rechunk_bsf: add bitstream filter to rechunk pcm audio

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Signed-off-by: Marton Balint <cus@passwd.hu>
This commit is contained in:
Marton Balint 2020-03-24 23:24:22 +01:00
parent d7a0071a44
commit 2035620b7c
6 changed files with 254 additions and 1 deletions
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1
Changelog
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@ -65,6 +65,7 @@ version <next>:
- Cunning Developments ADPCM decoder
- asubboost filter
- Pro Pinball Series Soundbank demuxer
- pcm_rechunk bitstream filter
version 4.2:

30
doc/bitstream_filters.texi
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@ -548,6 +548,36 @@ ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
@section null
This bitstream filter passes the packets through unchanged.
@section pcm_rechunk
Repacketize PCM audio to a fixed number of samples per packet or a fixed packet
rate per second. This is similar to the @ref{asetnsamples,,asetnsamples audio
filter,ffmpeg-filters} but works on audio packets instead of audio frames.
@table @option
@item nb_out_samples, n
Set the number of samples per each output audio packet. The number is intended
as the number of samples @emph{per each channel}. Default value is 1024.
@item pad, p
If set to 1, the filter will pad the last audio packet with silence, so that it
will contain the same number of samples (or roughly the same number of samples,
see @option{frame_rate}) as the previous ones. Default value is 1.
@item frame_rate, r
This option makes the filter output a fixed number of packets per second instead
of a fixed number of samples per packet. If the audio sample rate is not
divisible by the frame rate then the number of samples will not be constant but
will vary slightly so that each packet will start as close to the frame
boundary as possible. Using this option has precedence over @option{nb_out_samples}.
@end table
You can generate the well known 1602-1601-1602-1601-1602 pattern of 48kHz audio
for NTSC frame rate using the @option{frame_rate} option.
@example
ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -
@end example
@section prores_metadata
Modify color property metadata embedded in prores stream.

1
libavcodec/Makefile
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@ -1117,6 +1117,7 @@ OBJS-$(CONFIG_MPEG2_METADATA_BSF) += mpeg2_metadata_bsf.o
OBJS-$(CONFIG_NOISE_BSF) += noise_bsf.o
OBJS-$(CONFIG_NULL_BSF) += null_bsf.o
OBJS-$(CONFIG_OPUS_METADATA_BSF) += opus_metadata_bsf.o
OBJS-$(CONFIG_PCM_RECHUNK_BSF) += pcm_rechunk_bsf.o
OBJS-$(CONFIG_PRORES_METADATA_BSF) += prores_metadata_bsf.o
OBJS-$(CONFIG_REMOVE_EXTRADATA_BSF) += remove_extradata_bsf.o
OBJS-$(CONFIG_TEXT2MOVSUB_BSF) += movsub_bsf.o

1
libavcodec/bitstream_filters.c
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@ -50,6 +50,7 @@ extern const AVBitStreamFilter ff_mov2textsub_bsf;
extern const AVBitStreamFilter ff_noise_bsf;
extern const AVBitStreamFilter ff_null_bsf;
extern const AVBitStreamFilter ff_opus_metadata_bsf;
extern const AVBitStreamFilter ff_pcm_rechunk_bsf;
extern const AVBitStreamFilter ff_prores_metadata_bsf;
extern const AVBitStreamFilter ff_remove_extradata_bsf;
extern const AVBitStreamFilter ff_text2movsub_bsf;

220
libavcodec/pcm_rechunk_bsf.c Normal file
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@ -0,0 +1,220 @@
/*
* Copyright (c) 2020 Marton Balint
*
* 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 "avcodec.h"
#include "bsf.h"
#include "libavutil/avassert.h"
#include "libavutil/opt.h"
typedef struct PCMContext {
const AVClass *class;
int nb_out_samples;
int pad;
AVRational frame_rate;
AVPacket *in_pkt;
AVPacket *out_pkt;
int sample_size;
int64_t n;
} PCMContext;
static int init(AVBSFContext *ctx)
{
PCMContext *s = ctx->priv_data;
AVRational sr = av_make_q(ctx->par_in->sample_rate, 1);
int64_t min_samples;
if (ctx->par_in->channels <= 0 || ctx->par_in->sample_rate <= 0)
return AVERROR(EINVAL);
ctx->time_base_out = av_inv_q(sr);
s->sample_size = ctx->par_in->channels * av_get_bits_per_sample(ctx->par_in->codec_id) / 8;
if (s->frame_rate.num) {
min_samples = av_rescale_q_rnd(1, sr, s->frame_rate, AV_ROUND_DOWN);
} else {
min_samples = s->nb_out_samples;
}
if (min_samples <= 0 || min_samples > INT_MAX / s->sample_size - 1)
return AVERROR(EINVAL);
s->in_pkt = av_packet_alloc();
s->out_pkt = av_packet_alloc();
if (!s->in_pkt || !s->out_pkt)
return AVERROR(ENOMEM);
return 0;
}
static void uninit(AVBSFContext *ctx)
{
PCMContext *s = ctx->priv_data;
av_packet_free(&s->in_pkt);
av_packet_free(&s->out_pkt);
}
static void flush(AVBSFContext *ctx)
{
PCMContext *s = ctx->priv_data;
av_packet_unref(s->in_pkt);
av_packet_unref(s->out_pkt);
s->n = 0;
}
static int send_packet(PCMContext *s, int nb_samples, AVPacket *pkt)
{
pkt->duration = nb_samples;
s->n++;
return 0;
}
static void drain_packet(AVPacket *pkt, int drain_data, int drain_samples)
{
pkt->size -= drain_data;
pkt->data += drain_data;
if (pkt->dts != AV_NOPTS_VALUE)
pkt->dts += drain_samples;
if (pkt->pts != AV_NOPTS_VALUE)
pkt->pts += drain_samples;
}
static int get_next_nb_samples(AVBSFContext *ctx)
{
PCMContext *s = ctx->priv_data;
if (s->frame_rate.num) {
AVRational sr = av_make_q(ctx->par_in->sample_rate, 1);
return av_rescale_q(s->n + 1, sr, s->frame_rate) - av_rescale_q(s->n, sr, s->frame_rate);
} else {
return s->nb_out_samples;
}
}
static int rechunk_filter(AVBSFContext *ctx, AVPacket *pkt)
{
PCMContext *s = ctx->priv_data;
int nb_samples = get_next_nb_samples(ctx);
int data_size = nb_samples * s->sample_size;
int ret;
do {
if (s->in_pkt->size) {
if (s->out_pkt->size || s->in_pkt->size < data_size) {
int drain = FFMIN(s->in_pkt->size, data_size - s->out_pkt->size);
if (!s->out_pkt->size) {
ret = av_new_packet(s->out_pkt, data_size);
if (ret < 0)
return ret;
ret = av_packet_copy_props(s->out_pkt, s->in_pkt);
if (ret < 0) {
av_packet_unref(s->out_pkt);
return ret;
}
s->out_pkt->size = 0;
}
memcpy(s->out_pkt->data + s->out_pkt->size, s->in_pkt->data, drain);
s->out_pkt->size += drain;
drain_packet(s->in_pkt, drain, drain / s->sample_size);
if (!s->in_pkt->size)
av_packet_unref(s->in_pkt);
if (s->out_pkt->size == data_size) {
av_packet_move_ref(pkt, s->out_pkt);
return send_packet(s, nb_samples, pkt);
}
} else if (s->in_pkt->size > data_size) {
ret = av_packet_ref(pkt, s->in_pkt);
if (ret < 0)
return ret;
pkt->size = data_size;
drain_packet(s->in_pkt, data_size, nb_samples);
return send_packet(s, nb_samples, pkt);
} else {
av_assert0(s->in_pkt->size == data_size);
av_packet_move_ref(pkt, s->in_pkt);
return send_packet(s, nb_samples, pkt);
}
}
ret = ff_bsf_get_packet_ref(ctx, s->in_pkt);
if (ret == AVERROR_EOF && s->out_pkt->size) {
if (s->pad) {
memset(s->out_pkt->data + s->out_pkt->size, 0, data_size - s->out_pkt->size);
s->out_pkt->size = data_size;
} else {
nb_samples = s->out_pkt->size / s->sample_size;
}
av_packet_move_ref(pkt, s->out_pkt);
return send_packet(s, nb_samples, pkt);
}
if (ret >= 0)
av_packet_rescale_ts(s->in_pkt, ctx->time_base_in, ctx->time_base_out);
} while (ret >= 0);
return ret;
}
#define OFFSET(x) offsetof(PCMContext, x)
#define FLAGS (AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_BSF_PARAM)
static const AVOption options[] = {
{ "nb_out_samples", "set the number of per-packet output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS },
{ "n", "set the number of per-packet output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS },
{ "pad", "pad last packet with zeros", OFFSET(pad), AV_OPT_TYPE_BOOL, {.i64=1} , 0, 1, FLAGS },
{ "p", "pad last packet with zeros", OFFSET(pad), AV_OPT_TYPE_BOOL, {.i64=1} , 0, 1, FLAGS },
{ "frame_rate", "set number of packets per second", OFFSET(frame_rate), AV_OPT_TYPE_RATIONAL, {.dbl=0}, 0, INT_MAX, FLAGS },
{ "r", "set number of packets per second", OFFSET(frame_rate), AV_OPT_TYPE_RATIONAL, {.dbl=0}, 0, INT_MAX, FLAGS },
{ NULL },
};
static const AVClass pcm_rechunk_class = {
.class_name = "pcm_rechunk_bsf",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
static const enum AVCodecID codec_ids[] = {
AV_CODEC_ID_PCM_S16LE,
AV_CODEC_ID_PCM_S16BE,
AV_CODEC_ID_PCM_S8,
AV_CODEC_ID_PCM_S32LE,
AV_CODEC_ID_PCM_S32BE,
AV_CODEC_ID_PCM_S24LE,
AV_CODEC_ID_PCM_S24BE,
AV_CODEC_ID_PCM_F32BE,
AV_CODEC_ID_PCM_F32LE,
AV_CODEC_ID_PCM_F64BE,
AV_CODEC_ID_PCM_F64LE,
AV_CODEC_ID_PCM_S64LE,
AV_CODEC_ID_PCM_S64BE,
AV_CODEC_ID_PCM_F16LE,
AV_CODEC_ID_PCM_F24LE,
AV_CODEC_ID_NONE,
};
const AVBitStreamFilter ff_pcm_rechunk_bsf = {
.name = "pcm_rechunk",
.priv_data_size = sizeof(PCMContext),
.priv_class = &pcm_rechunk_class,
.filter = rechunk_filter,
.init = init,
.flush = flush,
.close = uninit,
.codec_ids = codec_ids,
};

2
libavcodec/version.h
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@ -28,7 +28,7 @@
#include "libavutil/version.h"
#define LIBAVCODEC_VERSION_MAJOR 58
#define LIBAVCODEC_VERSION_MINOR 82
#define LIBAVCODEC_VERSION_MINOR 83
#define LIBAVCODEC_VERSION_MICRO 100
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \