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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-12 19:18:44 +02:00
FFmpeg/libavformat/pcm.c
Marton Balint 44b2769619 avformat/pcm: decrease target audio frame per sec to 10
This makes the wav and pcm demuxer demux bigger packets, which is more
efficient.

As a side effect of the bigger packets, audio durations can become less exact
for command lines such as "ffmpeg -i $INPUT -c:a copy -t 1.0 $OUTPUT".

Signed-off-by: Marton Balint <cus@passwd.hu>
2024-03-16 19:19:42 +01:00

104 lines
3.5 KiB
C

/*
* PCM common functions
* Copyright (c) 2003 Fabrice Bellard
*
* 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 "libavutil/mathematics.h"
#include "avformat.h"
#include "internal.h"
#include "pcm.h"
#define PCM_DEMUX_TARGET_FPS 10
int ff_pcm_default_packet_size(AVCodecParameters *par)
{
int nb_samples, max_samples, bits_per_sample;
int64_t bitrate;
if (par->block_align <= 0)
return AVERROR(EINVAL);
max_samples = INT_MAX / par->block_align;
bits_per_sample = av_get_bits_per_sample(par->codec_id);
bitrate = par->bit_rate;
/* Don't trust the codecpar bitrate if we can calculate it ourselves */
if (bits_per_sample > 0 && par->sample_rate > 0 && par->ch_layout.nb_channels > 0)
if ((int64_t)par->sample_rate * par->ch_layout.nb_channels < INT64_MAX / bits_per_sample)
bitrate = bits_per_sample * par->sample_rate * par->ch_layout.nb_channels;
if (bitrate > 0) {
nb_samples = av_clip64(bitrate / 8 / PCM_DEMUX_TARGET_FPS / par->block_align, 1, max_samples);
nb_samples = 1 << av_log2(nb_samples);
} else {
/* Fallback to a size based method for a non-pcm codec with unknown bitrate */
nb_samples = av_clip(4096 / par->block_align, 1, max_samples);
}
return par->block_align * nb_samples;
}
int ff_pcm_read_packet(AVFormatContext *s, AVPacket *pkt)
{
int ret, size;
size = ff_pcm_default_packet_size(s->streams[0]->codecpar);
if (size < 0)
return size;
ret = av_get_packet(s->pb, pkt, size);
pkt->flags &= ~AV_PKT_FLAG_CORRUPT;
pkt->stream_index = 0;
return ret;
}
int ff_pcm_read_seek(AVFormatContext *s,
int stream_index, int64_t timestamp, int flags)
{
AVStream *st;
int block_align, byte_rate;
int64_t pos, ret;
st = s->streams[0];
block_align = st->codecpar->block_align ? st->codecpar->block_align :
(av_get_bits_per_sample(st->codecpar->codec_id) * st->codecpar->ch_layout.nb_channels) >> 3;
byte_rate = st->codecpar->bit_rate ? st->codecpar->bit_rate >> 3 :
block_align * st->codecpar->sample_rate;
if (block_align <= 0 || byte_rate <= 0)
return -1;
if (timestamp < 0) timestamp = 0;
/* compute the position by aligning it to block_align */
pos = av_rescale_rnd(timestamp * byte_rate,
st->time_base.num,
st->time_base.den * (int64_t)block_align,
(flags & AVSEEK_FLAG_BACKWARD) ? AV_ROUND_DOWN : AV_ROUND_UP);
pos *= block_align;
/* recompute exact position */
ffstream(st)->cur_dts = av_rescale(pos, st->time_base.den, byte_rate * (int64_t)st->time_base.num);
if ((ret = avio_seek(s->pb, pos + ffformatcontext(s)->data_offset, SEEK_SET)) < 0)
return ret;
return 0;
}