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https://github.com/FFmpeg/FFmpeg.git
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44b2769619
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>
104 lines
3.5 KiB
C
104 lines
3.5 KiB
C
/*
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* PCM common functions
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* Copyright (c) 2003 Fabrice Bellard
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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 "libavutil/mathematics.h"
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#include "avformat.h"
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#include "internal.h"
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#include "pcm.h"
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#define PCM_DEMUX_TARGET_FPS 10
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int ff_pcm_default_packet_size(AVCodecParameters *par)
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{
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int nb_samples, max_samples, bits_per_sample;
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int64_t bitrate;
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if (par->block_align <= 0)
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return AVERROR(EINVAL);
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max_samples = INT_MAX / par->block_align;
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bits_per_sample = av_get_bits_per_sample(par->codec_id);
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bitrate = par->bit_rate;
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/* Don't trust the codecpar bitrate if we can calculate it ourselves */
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if (bits_per_sample > 0 && par->sample_rate > 0 && par->ch_layout.nb_channels > 0)
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if ((int64_t)par->sample_rate * par->ch_layout.nb_channels < INT64_MAX / bits_per_sample)
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bitrate = bits_per_sample * par->sample_rate * par->ch_layout.nb_channels;
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if (bitrate > 0) {
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nb_samples = av_clip64(bitrate / 8 / PCM_DEMUX_TARGET_FPS / par->block_align, 1, max_samples);
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nb_samples = 1 << av_log2(nb_samples);
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} else {
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/* Fallback to a size based method for a non-pcm codec with unknown bitrate */
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nb_samples = av_clip(4096 / par->block_align, 1, max_samples);
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}
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return par->block_align * nb_samples;
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}
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int ff_pcm_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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int ret, size;
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size = ff_pcm_default_packet_size(s->streams[0]->codecpar);
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if (size < 0)
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return size;
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ret = av_get_packet(s->pb, pkt, size);
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pkt->flags &= ~AV_PKT_FLAG_CORRUPT;
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pkt->stream_index = 0;
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return ret;
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}
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int ff_pcm_read_seek(AVFormatContext *s,
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int stream_index, int64_t timestamp, int flags)
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{
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AVStream *st;
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int block_align, byte_rate;
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int64_t pos, ret;
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st = s->streams[0];
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block_align = st->codecpar->block_align ? st->codecpar->block_align :
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(av_get_bits_per_sample(st->codecpar->codec_id) * st->codecpar->ch_layout.nb_channels) >> 3;
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byte_rate = st->codecpar->bit_rate ? st->codecpar->bit_rate >> 3 :
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block_align * st->codecpar->sample_rate;
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if (block_align <= 0 || byte_rate <= 0)
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return -1;
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if (timestamp < 0) timestamp = 0;
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/* compute the position by aligning it to block_align */
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pos = av_rescale_rnd(timestamp * byte_rate,
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st->time_base.num,
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st->time_base.den * (int64_t)block_align,
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(flags & AVSEEK_FLAG_BACKWARD) ? AV_ROUND_DOWN : AV_ROUND_UP);
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pos *= block_align;
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/* recompute exact position */
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ffstream(st)->cur_dts = av_rescale(pos, st->time_base.den, byte_rate * (int64_t)st->time_base.num);
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if ((ret = avio_seek(s->pb, pos + ffformatcontext(s)->data_offset, SEEK_SET)) < 0)
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return ret;
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return 0;
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}
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