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https://github.com/FFmpeg/FFmpeg.git
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f8911b987d
* qatar/master: mss3: use standard zigzag table mss3: split DSP functions that are used in MTS2(MSS4) into separate file motion-test: do not use getopt() tcp: add initial timeout limit for incoming connections configure: Change the rdtsc check to a linker check avconv: propagate fatal errors from lavfi. lavfi: add error handling to filter_samples(). fate-run: make avconv() properly deal with multiple inputs. asplit: don't leak the input buffer. af_resample: fix request_frame() behavior. af_asyncts: fix request_frame() behavior. libx264: support aspect ratio switching matroskadec: honor error_recognition when encountering unknown elements. lavr: resampling: add support for s32p, fltp, and dblp internal sample formats lavr: resampling: add filter type and Kaiser window beta to AVOptions lavr: Use AV_SAMPLE_FMT_NONE to auto-select the internal sample format lavr: mix: validate internal sample format in ff_audio_mix_init() Conflicts: ffmpeg.c ffplay.c libavcodec/libx264.c libavfilter/audio.c libavfilter/split.c libavformat/tcp.c tests/fate-run.sh Merged-by: Michael Niedermayer <michaelni@gmx.at>
107 lines
3.8 KiB
C
107 lines
3.8 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 "libavutil/adler32.h"
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#include "libavutil/audioconvert.h"
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#include "libavutil/timestamp.h"
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#include "audio.h"
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#include "avfilter.h"
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typedef struct {
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unsigned int frame;
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} ShowInfoContext;
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static av_cold int init(AVFilterContext *ctx, const char *args)
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{
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ShowInfoContext *showinfo = ctx->priv;
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showinfo->frame = 0;
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return 0;
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}
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static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
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{
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AVFilterContext *ctx = inlink->dst;
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ShowInfoContext *showinfo = ctx->priv;
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uint32_t plane_checksum[8] = {0}, checksum = 0;
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char chlayout_str[128];
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int plane;
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int linesize =
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samplesref->audio->nb_samples *
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av_get_bytes_per_sample(samplesref->format);
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if (!av_sample_fmt_is_planar(samplesref->format))
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linesize *= av_get_channel_layout_nb_channels(samplesref->audio->channel_layout);
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for (plane = 0; samplesref->data[plane] && plane < 8; plane++) {
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uint8_t *data = samplesref->data[plane];
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plane_checksum[plane] = av_adler32_update(plane_checksum[plane],
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data, linesize);
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checksum = av_adler32_update(checksum, data, linesize);
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}
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av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str), -1,
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samplesref->audio->channel_layout);
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av_log(ctx, AV_LOG_INFO,
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"n:%d pts:%s pts_time:%s pos:%"PRId64" "
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"fmt:%s chlayout:%s nb_samples:%d rate:%d "
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"checksum:%08X plane_checksum[%08X",
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showinfo->frame,
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av_ts2str(samplesref->pts), av_ts2timestr(samplesref->pts, &inlink->time_base),
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samplesref->pos,
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av_get_sample_fmt_name(samplesref->format),
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chlayout_str,
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samplesref->audio->nb_samples,
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samplesref->audio->sample_rate,
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checksum,
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plane_checksum[0]);
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for (plane = 1; samplesref->data[plane] && plane < 8; plane++)
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av_log(ctx, AV_LOG_INFO, " %08X", plane_checksum[plane]);
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av_log(ctx, AV_LOG_INFO, "]\n");
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showinfo->frame++;
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return ff_filter_samples(inlink->dst->outputs[0], samplesref);
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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(ShowInfoContext),
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.init = init,
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.inputs = (const AVFilterPad[]) {{ .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_samples = filter_samples,
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.min_perms = AV_PERM_READ, },
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{ .name = NULL}},
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.outputs = (const AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO },
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{ .name = NULL}},
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};
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