mirror of
https://github.com/FFmpeg/FFmpeg.git
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275 lines
9.5 KiB
C
275 lines
9.5 KiB
C
/*
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* Copyright (c) 2011 Stefano Sabatini
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*
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* This file is part of Libav.
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*
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* Libav 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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* Libav 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 Libav; 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/attributes.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "libavutil/downmix_info.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/mem.h"
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#include "libavutil/replaygain.h"
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#include "libavutil/samplefmt.h"
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#include "libavcodec/avcodec.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 int config_input(AVFilterLink *inlink)
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{
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AShowInfoContext *s = inlink->dst->priv;
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int channels = av_get_channel_layout_nb_channels(inlink->channel_layout);
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s->plane_checksums = av_malloc(channels * sizeof(*s->plane_checksums));
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if (!s->plane_checksums)
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return AVERROR(ENOMEM);
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return 0;
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}
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static av_cold 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 void dump_matrixenc(AVFilterContext *ctx, AVFrameSideData *sd)
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{
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enum AVMatrixEncoding enc;
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av_log(ctx, AV_LOG_INFO, "matrix encoding: ");
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if (sd->size < sizeof(enum AVMatrixEncoding)) {
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av_log(ctx, AV_LOG_INFO, "invalid data");
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return;
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}
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enc = *(enum AVMatrixEncoding *)sd->data;
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switch (enc) {
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case AV_MATRIX_ENCODING_NONE: av_log(ctx, AV_LOG_INFO, "none"); break;
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case AV_MATRIX_ENCODING_DOLBY: av_log(ctx, AV_LOG_INFO, "Dolby Surround"); break;
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case AV_MATRIX_ENCODING_DPLII: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic II"); break;
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case AV_MATRIX_ENCODING_DPLIIX: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic IIx"); break;
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case AV_MATRIX_ENCODING_DPLIIZ: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic IIz"); break;
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case AV_MATRIX_ENCODING_DOLBYEX: av_log(ctx, AV_LOG_INFO, "Dolby EX"); break;
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case AV_MATRIX_ENCODING_DOLBYHEADPHONE: av_log(ctx, AV_LOG_INFO, "Dolby Headphone"); break;
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default: av_log(ctx, AV_LOG_WARNING, "unknown"); break;
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}
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}
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static void dump_downmix(AVFilterContext *ctx, AVFrameSideData *sd)
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{
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AVDownmixInfo *di;
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av_log(ctx, AV_LOG_INFO, "downmix: ");
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if (sd->size < sizeof(*di)) {
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av_log(ctx, AV_LOG_INFO, "invalid data");
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return;
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}
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di = (AVDownmixInfo *)sd->data;
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av_log(ctx, AV_LOG_INFO, "preferred downmix type - ");
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switch (di->preferred_downmix_type) {
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case AV_DOWNMIX_TYPE_LORO: av_log(ctx, AV_LOG_INFO, "Lo/Ro"); break;
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case AV_DOWNMIX_TYPE_LTRT: av_log(ctx, AV_LOG_INFO, "Lt/Rt"); break;
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case AV_DOWNMIX_TYPE_DPLII: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic II"); break;
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default: av_log(ctx, AV_LOG_WARNING, "unknown"); break;
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}
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av_log(ctx, AV_LOG_INFO, " Mix levels: center %f (%f ltrt) - "
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"surround %f (%f ltrt) - lfe %f",
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di->center_mix_level, di->center_mix_level_ltrt,
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di->surround_mix_level, di->surround_mix_level_ltrt,
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di->lfe_mix_level);
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}
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static void print_gain(AVFilterContext *ctx, const char *str, int32_t gain)
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{
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av_log(ctx, AV_LOG_INFO, "%s - ", str);
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if (gain == INT32_MIN)
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av_log(ctx, AV_LOG_INFO, "unknown");
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else
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av_log(ctx, AV_LOG_INFO, "%f", gain / 100000.0f);
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av_log(ctx, AV_LOG_INFO, ", ");
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}
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static void print_peak(AVFilterContext *ctx, const char *str, uint32_t peak)
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{
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av_log(ctx, AV_LOG_INFO, "%s - ", str);
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if (!peak)
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av_log(ctx, AV_LOG_INFO, "unknown");
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else
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av_log(ctx, AV_LOG_INFO, "%f", (float)peak / UINT32_MAX);
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av_log(ctx, AV_LOG_INFO, ", ");
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}
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static void dump_replaygain(AVFilterContext *ctx, AVFrameSideData *sd)
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{
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AVReplayGain *rg;
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av_log(ctx, AV_LOG_INFO, "replaygain: ");
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if (sd->size < sizeof(*rg)) {
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av_log(ctx, AV_LOG_INFO, "invalid data");
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return;
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}
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rg = (AVReplayGain*)sd->data;
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print_gain(ctx, "track gain", rg->track_gain);
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print_peak(ctx, "track peak", rg->track_peak);
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print_gain(ctx, "album gain", rg->album_gain);
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print_peak(ctx, "album peak", rg->album_peak);
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}
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static void dump_audio_service_type(AVFilterContext *ctx, AVFrameSideData *sd)
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{
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enum AVAudioServiceType *ast;
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av_log(ctx, AV_LOG_INFO, "audio service type: ");
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if (sd->size < sizeof(*ast)) {
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av_log(ctx, AV_LOG_INFO, "invalid data");
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return;
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}
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ast = (enum AVAudioServiceType*)sd->data;
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switch (*ast) {
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case AV_AUDIO_SERVICE_TYPE_MAIN: av_log(ctx, AV_LOG_INFO, "Main Audio Service"); break;
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case AV_AUDIO_SERVICE_TYPE_EFFECTS: av_log(ctx, AV_LOG_INFO, "Effects"); break;
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case AV_AUDIO_SERVICE_TYPE_VISUALLY_IMPAIRED: av_log(ctx, AV_LOG_INFO, "Visually Impaired"); break;
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case AV_AUDIO_SERVICE_TYPE_HEARING_IMPAIRED: av_log(ctx, AV_LOG_INFO, "Hearing Impaired"); break;
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case AV_AUDIO_SERVICE_TYPE_DIALOGUE: av_log(ctx, AV_LOG_INFO, "Dialogue"); break;
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case AV_AUDIO_SERVICE_TYPE_COMMENTARY: av_log(ctx, AV_LOG_INFO, "Commentary"); break;
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case AV_AUDIO_SERVICE_TYPE_EMERGENCY: av_log(ctx, AV_LOG_INFO, "Emergency"); break;
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case AV_AUDIO_SERVICE_TYPE_VOICE_OVER: av_log(ctx, AV_LOG_INFO, "Voice Over"); break;
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case AV_AUDIO_SERVICE_TYPE_KARAOKE: av_log(ctx, AV_LOG_INFO, "Karaoke"); break;
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default: av_log(ctx, AV_LOG_INFO, "unknown"); break;
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}
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}
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static void dump_unknown(AVFilterContext *ctx, AVFrameSideData *sd)
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{
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av_log(ctx, AV_LOG_INFO, "unknown side data type: %d, size %d bytes", sd->type, sd->size);
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *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->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->nb_samples * block_align;
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int planes = planar ? channels : 1;
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int i;
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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->channel_layout);
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av_log(ctx, AV_LOG_INFO,
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"n:%"PRIu64" pts:%"PRId64" pts_time:%f "
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"fmt:%s chlayout:%s rate:%d nb_samples:%d "
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"checksum:%08"PRIX32" ",
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s->frame, buf->pts, buf->pts * av_q2d(inlink->time_base),
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av_get_sample_fmt_name(buf->format), chlayout_str,
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buf->sample_rate, buf->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, "%08"PRIX32" ", s->plane_checksums[i]);
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av_log(ctx, AV_LOG_INFO, "]\n");
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for (i = 0; i < buf->nb_side_data; i++) {
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AVFrameSideData *sd = buf->side_data[i];
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av_log(ctx, AV_LOG_INFO, " side data - ");
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switch (sd->type) {
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case AV_FRAME_DATA_MATRIXENCODING: dump_matrixenc (ctx, sd); break;
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case AV_FRAME_DATA_DOWNMIX_INFO: dump_downmix (ctx, sd); break;
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case AV_FRAME_DATA_REPLAYGAIN: dump_replaygain(ctx, sd); break;
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case AV_FRAME_DATA_AUDIO_SERVICE_TYPE: dump_audio_service_type(ctx, sd); break;
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default: dump_unknown (ctx, sd); break;
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}
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av_log(ctx, AV_LOG_INFO, "\n");
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}
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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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.config_props = config_input,
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.filter_frame = filter_frame,
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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 ff_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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