87b9dc0982
The commit has already been split and commited, this merge is just for metadata. What has not been merged is the removial of some start/end_frame / draw_slice related functions that are still in use. They can be removed once they are unused. * commit '565e4993c63f797e2d50ad2f1e8f62fdbe299666': lavfi: merge start_frame/draw_slice/end_frame Merged-by: Michael Niedermayer <michaelni@gmx.at>
134 lines
4.0 KiB
C
134 lines
4.0 KiB
C
/*
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* Copyright (c) 2011 Stefano Sabatini
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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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/**
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* @file
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* filter for showing textual audio frame information
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*/
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#include <inttypes.h>
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#include <stddef.h>
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#include "libavutil/adler32.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "libavutil/mem.h"
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#include "libavutil/timestamp.h"
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#include "libavutil/samplefmt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct AShowInfoContext {
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/**
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* Scratch space for individual plane checksums for planar audio
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*/
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uint32_t *plane_checksums;
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/**
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* Frame counter
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*/
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uint64_t frame;
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} AShowInfoContext;
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static void uninit(AVFilterContext *ctx)
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{
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AShowInfoContext *s = ctx->priv;
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av_freep(&s->plane_checksums);
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}
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static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
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{
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AVFilterContext *ctx = inlink->dst;
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AShowInfoContext *s = ctx->priv;
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char chlayout_str[128];
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uint32_t checksum = 0;
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int channels = av_get_channel_layout_nb_channels(buf->audio->channel_layout);
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int planar = av_sample_fmt_is_planar(buf->format);
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int block_align = av_get_bytes_per_sample(buf->format) * (planar ? 1 : channels);
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int data_size = buf->audio->nb_samples * block_align;
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int planes = planar ? channels : 1;
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int i;
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void *tmp_ptr = av_realloc(s->plane_checksums, channels * sizeof(*s->plane_checksums));
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if (!tmp_ptr)
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return AVERROR(ENOMEM);
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s->plane_checksums = tmp_ptr;
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for (i = 0; i < planes; i++) {
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uint8_t *data = buf->extended_data[i];
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s->plane_checksums[i] = av_adler32_update(0, data, data_size);
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checksum = i ? av_adler32_update(checksum, data, data_size) :
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s->plane_checksums[0];
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}
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av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str), -1,
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buf->audio->channel_layout);
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av_log(ctx, AV_LOG_INFO,
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"n:%"PRIu64" pts:%s pts_time:%s pos:%"PRId64" "
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"fmt:%s channels:%d chlayout:%s rate:%d nb_samples:%d "
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"checksum:%08X ",
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s->frame,
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av_ts2str(buf->pts), av_ts2timestr(buf->pts, &inlink->time_base),
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buf->pos,
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av_get_sample_fmt_name(buf->format), buf->audio->channels, chlayout_str,
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buf->audio->sample_rate, buf->audio->nb_samples,
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checksum);
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av_log(ctx, AV_LOG_INFO, "plane_checksums: [ ");
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for (i = 0; i < planes; i++)
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av_log(ctx, AV_LOG_INFO, "%08X ", s->plane_checksums[i]);
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av_log(ctx, AV_LOG_INFO, "]\n");
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s->frame++;
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return ff_filter_frame(inlink->dst->outputs[0], buf);
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}
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static const AVFilterPad inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.get_audio_buffer = ff_null_get_audio_buffer,
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.filter_frame = filter_frame,
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.min_perms = AV_PERM_READ,
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},
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{ NULL },
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};
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static const AVFilterPad outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{ NULL },
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};
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AVFilter avfilter_af_ashowinfo = {
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.name = "ashowinfo",
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.description = NULL_IF_CONFIG_SMALL("Show textual information for each audio frame."),
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.priv_size = sizeof(AShowInfoContext),
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.uninit = uninit,
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.inputs = inputs,
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.outputs = outputs,
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};
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