mirror of
https://git.ffmpeg.org/ffmpeg.git
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242 lines
8.0 KiB
C
242 lines
8.0 KiB
C
/*
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* Copyright (c) 2020 Marton Balint
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "bsf.h"
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#include "bsf_internal.h"
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#include "libavutil/avassert.h"
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#include "libavutil/opt.h"
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typedef struct PCMContext {
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const AVClass *class;
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int nb_out_samples;
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int pad;
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AVRational frame_rate;
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AVPacket *in_pkt;
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AVPacket *out_pkt;
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int sample_size;
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int64_t n;
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} PCMContext;
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static int init(AVBSFContext *ctx)
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{
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PCMContext *s = ctx->priv_data;
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AVRational sr = av_make_q(ctx->par_in->sample_rate, 1);
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int64_t min_samples;
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if (ctx->par_in->ch_layout.nb_channels <= 0 || ctx->par_in->sample_rate <= 0)
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return AVERROR(EINVAL);
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ctx->time_base_out = av_inv_q(sr);
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s->sample_size = ctx->par_in->ch_layout.nb_channels *
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av_get_bits_per_sample(ctx->par_in->codec_id) / 8;
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if (s->frame_rate.num) {
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min_samples = av_rescale_q_rnd(1, sr, s->frame_rate, AV_ROUND_DOWN);
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} else {
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min_samples = s->nb_out_samples;
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}
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if (min_samples <= 0 || min_samples > INT_MAX / s->sample_size - 1)
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return AVERROR(EINVAL);
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s->in_pkt = av_packet_alloc();
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s->out_pkt = av_packet_alloc();
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if (!s->in_pkt || !s->out_pkt)
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return AVERROR(ENOMEM);
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return 0;
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}
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static void uninit(AVBSFContext *ctx)
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{
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PCMContext *s = ctx->priv_data;
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av_packet_free(&s->in_pkt);
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av_packet_free(&s->out_pkt);
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}
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static void flush(AVBSFContext *ctx)
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{
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PCMContext *s = ctx->priv_data;
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av_packet_unref(s->in_pkt);
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av_packet_unref(s->out_pkt);
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s->n = 0;
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}
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static int send_packet(PCMContext *s, int nb_samples, AVPacket *pkt)
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{
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pkt->duration = nb_samples;
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s->n++;
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return 0;
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}
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static void drain_packet(AVPacket *pkt, int drain_data, int drain_samples)
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{
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pkt->size -= drain_data;
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pkt->data += drain_data;
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if (pkt->dts != AV_NOPTS_VALUE)
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pkt->dts += drain_samples;
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if (pkt->pts != AV_NOPTS_VALUE)
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pkt->pts += drain_samples;
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}
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static int get_next_nb_samples(AVBSFContext *ctx)
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{
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PCMContext *s = ctx->priv_data;
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if (s->frame_rate.num) {
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AVRational sr = av_make_q(ctx->par_in->sample_rate, 1);
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return av_rescale_q(s->n + 1, sr, s->frame_rate) - av_rescale_q(s->n, sr, s->frame_rate);
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} else {
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return s->nb_out_samples;
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}
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}
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static void set_silence(AVCodecParameters *par, uint8_t *buf, size_t size)
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{
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int c = 0;
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switch (par->codec_id) {
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case AV_CODEC_ID_PCM_ALAW: c = 0xd5; break;
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case AV_CODEC_ID_PCM_MULAW:
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case AV_CODEC_ID_PCM_VIDC: c = 0xff; break;
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case AV_CODEC_ID_PCM_U8: c = 0x80; break;
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}
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memset(buf, c, size);
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}
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static int rechunk_filter(AVBSFContext *ctx, AVPacket *pkt)
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{
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PCMContext *s = ctx->priv_data;
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int nb_samples = get_next_nb_samples(ctx);
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int data_size = nb_samples * s->sample_size;
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int ret;
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do {
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if (s->in_pkt->size) {
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if (s->out_pkt->size || s->in_pkt->size < data_size) {
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int drain = FFMIN(s->in_pkt->size, data_size - s->out_pkt->size);
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if (!s->out_pkt->size) {
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ret = av_new_packet(s->out_pkt, data_size);
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if (ret < 0)
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return ret;
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ret = av_packet_copy_props(s->out_pkt, s->in_pkt);
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if (ret < 0) {
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av_packet_unref(s->out_pkt);
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return ret;
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}
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s->out_pkt->size = 0;
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}
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memcpy(s->out_pkt->data + s->out_pkt->size, s->in_pkt->data, drain);
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s->out_pkt->size += drain;
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drain_packet(s->in_pkt, drain, drain / s->sample_size);
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if (!s->in_pkt->size)
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av_packet_unref(s->in_pkt);
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if (s->out_pkt->size == data_size) {
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av_packet_move_ref(pkt, s->out_pkt);
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return send_packet(s, nb_samples, pkt);
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}
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av_assert0(!s->in_pkt->size);
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} else if (s->in_pkt->size > data_size) {
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ret = av_packet_ref(pkt, s->in_pkt);
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if (ret < 0)
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return ret;
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pkt->size = data_size;
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drain_packet(s->in_pkt, data_size, nb_samples);
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return send_packet(s, nb_samples, pkt);
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} else {
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av_assert0(s->in_pkt->size == data_size);
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av_packet_move_ref(pkt, s->in_pkt);
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return send_packet(s, nb_samples, pkt);
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}
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} else
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av_packet_unref(s->in_pkt);
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ret = ff_bsf_get_packet_ref(ctx, s->in_pkt);
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if (ret == AVERROR_EOF && s->out_pkt->size) {
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if (s->pad) {
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set_silence(ctx->par_in, s->out_pkt->data + s->out_pkt->size, data_size - s->out_pkt->size);
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s->out_pkt->size = data_size;
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} else {
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nb_samples = s->out_pkt->size / s->sample_size;
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}
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av_packet_move_ref(pkt, s->out_pkt);
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return send_packet(s, nb_samples, pkt);
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}
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if (ret >= 0)
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av_packet_rescale_ts(s->in_pkt, ctx->time_base_in, ctx->time_base_out);
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} while (ret >= 0);
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return ret;
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}
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#define OFFSET(x) offsetof(PCMContext, x)
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#define FLAGS (AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_BSF_PARAM)
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static const AVOption options[] = {
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{ "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 },
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{ "n", "set the number of per-packet output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS },
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{ "pad", "pad last packet with zeros", OFFSET(pad), AV_OPT_TYPE_BOOL, {.i64=1} , 0, 1, FLAGS },
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{ "p", "pad last packet with zeros", OFFSET(pad), AV_OPT_TYPE_BOOL, {.i64=1} , 0, 1, FLAGS },
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{ "frame_rate", "set number of packets per second", OFFSET(frame_rate), AV_OPT_TYPE_RATIONAL, {.dbl=0}, 0, INT_MAX, FLAGS },
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{ "r", "set number of packets per second", OFFSET(frame_rate), AV_OPT_TYPE_RATIONAL, {.dbl=0}, 0, INT_MAX, FLAGS },
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{ NULL },
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};
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static const AVClass pcm_rechunk_class = {
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.class_name = "pcm_rechunk_bsf",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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static const enum AVCodecID codec_ids[] = {
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AV_CODEC_ID_PCM_ALAW,
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AV_CODEC_ID_PCM_F16LE,
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AV_CODEC_ID_PCM_F24LE,
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AV_CODEC_ID_PCM_F32BE,
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AV_CODEC_ID_PCM_F32LE,
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AV_CODEC_ID_PCM_F64BE,
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AV_CODEC_ID_PCM_F64LE,
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AV_CODEC_ID_PCM_MULAW,
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AV_CODEC_ID_PCM_S16BE,
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AV_CODEC_ID_PCM_S16LE,
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AV_CODEC_ID_PCM_S24BE,
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AV_CODEC_ID_PCM_S24DAUD,
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AV_CODEC_ID_PCM_S24LE,
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AV_CODEC_ID_PCM_S32BE,
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AV_CODEC_ID_PCM_S32LE,
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AV_CODEC_ID_PCM_S64BE,
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AV_CODEC_ID_PCM_S64LE,
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AV_CODEC_ID_PCM_S8,
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AV_CODEC_ID_PCM_SGA,
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AV_CODEC_ID_PCM_U8,
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AV_CODEC_ID_PCM_VIDC,
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AV_CODEC_ID_NONE,
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};
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const FFBitStreamFilter ff_pcm_rechunk_bsf = {
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.p.name = "pcm_rechunk",
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.p.codec_ids = codec_ids,
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.p.priv_class = &pcm_rechunk_class,
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.priv_data_size = sizeof(PCMContext),
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.filter = rechunk_filter,
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.init = init,
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.flush = flush,
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.close = uninit,
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};
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