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FFmpeg/libavfilter/af_ashowinfo.c
2014-03-24 06:14:55 +01:00

248 lines
8.0 KiB
C

/*
* Copyright (c) 2011 Stefano Sabatini
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* filter for showing textual audio frame information
*/
#include <inttypes.h>
#include <stddef.h>
#include "libavutil/adler32.h"
#include "libavutil/attributes.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/downmix_info.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/mem.h"
#include "libavutil/replaygain.h"
#include "libavutil/samplefmt.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"
typedef struct AShowInfoContext {
/**
* Scratch space for individual plane checksums for planar audio
*/
uint32_t *plane_checksums;
/**
* Frame counter
*/
uint64_t frame;
} AShowInfoContext;
static int config_input(AVFilterLink *inlink)
{
AShowInfoContext *s = inlink->dst->priv;
int channels = av_get_channel_layout_nb_channels(inlink->channel_layout);
s->plane_checksums = av_malloc(channels * sizeof(*s->plane_checksums));
if (!s->plane_checksums)
return AVERROR(ENOMEM);
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AShowInfoContext *s = ctx->priv;
av_freep(&s->plane_checksums);
}
static void dump_matrixenc(AVFilterContext *ctx, AVFrameSideData *sd)
{
enum AVMatrixEncoding enc;
av_log(ctx, AV_LOG_INFO, "matrix encoding: ");
if (sd->size < sizeof(enum AVMatrixEncoding)) {
av_log(ctx, AV_LOG_INFO, "invalid data");
return;
}
enc = *(enum AVMatrixEncoding *)sd->data;
switch (enc) {
case AV_MATRIX_ENCODING_NONE: av_log(ctx, AV_LOG_INFO, "none"); break;
case AV_MATRIX_ENCODING_DOLBY: av_log(ctx, AV_LOG_INFO, "Dolby"); break;
case AV_MATRIX_ENCODING_DPLII: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic II"); break;
case AV_MATRIX_ENCODING_DPLIIX: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic IIx"); break;
case AV_MATRIX_ENCODING_DPLIIZ: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic IIz"); break;
case AV_MATRIX_ENCODING_DOLBYEX: av_log(ctx, AV_LOG_INFO, "Dolby EX"); break;
case AV_MATRIX_ENCODING_DOLBYHEADPHONE: av_log(ctx, AV_LOG_INFO, "Dolby Headphone"); break;
default: av_log(ctx, AV_LOG_WARNING, "unknown"); break;
}
}
static void dump_downmix(AVFilterContext *ctx, AVFrameSideData *sd)
{
AVDownmixInfo *di;
av_log(ctx, AV_LOG_INFO, "downmix: ");
if (sd->size < sizeof(*di)) {
av_log(ctx, AV_LOG_INFO, "invalid data");
return;
}
di = (AVDownmixInfo *)sd->data;
av_log(ctx, AV_LOG_INFO, "preferred downmix type - ");
switch (di->preferred_downmix_type) {
case AV_DOWNMIX_TYPE_LORO: av_log(ctx, AV_LOG_INFO, "Lo/Ro"); break;
case AV_DOWNMIX_TYPE_LTRT: av_log(ctx, AV_LOG_INFO, "Lt/Rt"); break;
case AV_DOWNMIX_TYPE_DPLII: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic II"); break;
default: av_log(ctx, AV_LOG_WARNING, "unknown"); break;
}
av_log(ctx, AV_LOG_INFO, " Mix levels: center %f (%f ltrt) - "
"surround %f (%f ltrt) - lfe %f",
di->center_mix_level, di->center_mix_level_ltrt,
di->surround_mix_level, di->surround_mix_level_ltrt,
di->lfe_mix_level);
}
static void print_gain(AVFilterContext *ctx, const char *str, int32_t gain)
{
av_log(ctx, AV_LOG_INFO, "%s - ", str);
if (gain == INT32_MIN)
av_log(ctx, AV_LOG_INFO, "unknown");
else
av_log(ctx, AV_LOG_INFO, "%f", gain / 100000.0f);
av_log(ctx, AV_LOG_INFO, ", ");
}
static void print_peak(AVFilterContext *ctx, const char *str, uint32_t peak)
{
av_log(ctx, AV_LOG_INFO, "%s - ", str);
if (!peak)
av_log(ctx, AV_LOG_INFO, "unknown");
else
av_log(ctx, AV_LOG_INFO, "%f", (float)peak / UINT32_MAX);
av_log(ctx, AV_LOG_INFO, ", ");
}
static void dump_replaygain(AVFilterContext *ctx, AVFrameSideData *sd)
{
AVReplayGain *rg;
av_log(ctx, AV_LOG_INFO, "replaygain: ");
if (sd->size < sizeof(*rg)) {
av_log(ctx, AV_LOG_INFO, "invalid data");
return;
}
rg = (AVReplayGain*)sd->data;
print_gain(ctx, "track gain", rg->track_gain);
print_peak(ctx, "track peak", rg->track_peak);
print_gain(ctx, "album gain", rg->album_gain);
print_peak(ctx, "album peak", rg->album_peak);
}
static void dump_unknown(AVFilterContext *ctx, AVFrameSideData *sd)
{
av_log(ctx, AV_LOG_INFO, "unknown side data type: %d, size %d bytes", sd->type, sd->size);
}
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
{
AVFilterContext *ctx = inlink->dst;
AShowInfoContext *s = ctx->priv;
char chlayout_str[128];
uint32_t checksum = 0;
int channels = av_get_channel_layout_nb_channels(buf->channel_layout);
int planar = av_sample_fmt_is_planar(buf->format);
int block_align = av_get_bytes_per_sample(buf->format) * (planar ? 1 : channels);
int data_size = buf->nb_samples * block_align;
int planes = planar ? channels : 1;
int i;
for (i = 0; i < planes; i++) {
uint8_t *data = buf->extended_data[i];
s->plane_checksums[i] = av_adler32_update(0, data, data_size);
checksum = i ? av_adler32_update(checksum, data, data_size) :
s->plane_checksums[0];
}
av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str), -1,
buf->channel_layout);
av_log(ctx, AV_LOG_INFO,
"n:%"PRIu64" pts:%"PRId64" pts_time:%f "
"fmt:%s chlayout:%s rate:%d nb_samples:%d "
"checksum:%08"PRIX32" ",
s->frame, buf->pts, buf->pts * av_q2d(inlink->time_base),
av_get_sample_fmt_name(buf->format), chlayout_str,
buf->sample_rate, buf->nb_samples,
checksum);
av_log(ctx, AV_LOG_INFO, "plane_checksums: [ ");
for (i = 0; i < planes; i++)
av_log(ctx, AV_LOG_INFO, "%08"PRIX32" ", s->plane_checksums[i]);
av_log(ctx, AV_LOG_INFO, "]\n");
for (i = 0; i < buf->nb_side_data; i++) {
AVFrameSideData *sd = buf->side_data[i];
av_log(ctx, AV_LOG_INFO, " side data - ");
switch (sd->type) {
case AV_FRAME_DATA_MATRIXENCODING: dump_matrixenc (ctx, sd); break;
case AV_FRAME_DATA_DOWNMIX_INFO: dump_downmix (ctx, sd); break;
case AV_FRAME_DATA_REPLAYGAIN: dump_replaygain(ctx, sd); break;
default: dump_unknown (ctx, sd); break;
}
av_log(ctx, AV_LOG_INFO, "\n");
}
s->frame++;
return ff_filter_frame(inlink->dst->outputs[0], buf);
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.get_audio_buffer = ff_null_get_audio_buffer,
.config_props = config_input,
.filter_frame = filter_frame,
},
{ NULL },
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL },
};
AVFilter ff_af_ashowinfo = {
.name = "ashowinfo",
.description = NULL_IF_CONFIG_SMALL("Show textual information for each audio frame."),
.priv_size = sizeof(AShowInfoContext),
.uninit = uninit,
.inputs = inputs,
.outputs = outputs,
};