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FFmpeg/libavcodec/pcm_rechunk_bsf.c
James Almer 86645bf0a6 avcodec/pcm_rechunk_bsf: assert that in_pkt is empty after merging its data into out_pkt
If it's not empty here, then a leak would ocurr immediately after.

Signed-off-by: James Almer <jamrial@gmail.com>
2023-04-21 17:14:36 -03:00

224 lines
7.5 KiB
C

/*
* 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 "bsf.h"
#include "bsf_internal.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->ch_layout.nb_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->ch_layout.nb_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);
}
av_assert0(!s->in_pkt->size);
} 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);
}
} else
av_packet_unref(s->in_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 FFBitStreamFilter ff_pcm_rechunk_bsf = {
.p.name = "pcm_rechunk",
.p.codec_ids = codec_ids,
.p.priv_class = &pcm_rechunk_class,
.priv_data_size = sizeof(PCMContext),
.filter = rechunk_filter,
.init = init,
.flush = flush,
.close = uninit,
};